text: /*
   * Copyright (c) 2017 Paul B Mahol
   *
   * This file is part of FFmpeg.
   *
   * FFmpeg is free software; you can redistribute it and/or
   * modify it under the terms of the GNU Lesser General Public
   * License as published by the Free Software Foundation; either
   * version 2.1 of the License, or (at your option) any later version.
   *
   * FFmpeg is distributed in the hope that it will be useful,
   * but WITHOUT ANY WARRANTY; without even the implied warranty of
   * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
   * Lesser General Public License for more details.
   *
   * You should have received a copy of the GNU Lesser General Public
   * License along with FFmpeg; if not, write to the Free Software
   * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
   */
  
  /**
   * @file
   * An arbitrary audio FIR filter
   */
  
  #include <float.h>
  
  #include "libavutil/avstring.h"
  #include "libavutil/channel_layout.h"
  #include "libavutil/common.h"
  #include "libavutil/float_dsp.h"
  #include "libavutil/intreadwrite.h"
  #include "libavutil/opt.h"
  #include "libavutil/xga_font_data.h"
  #include "libavcodec/avfft.h"
  
  #include "audio.h"
  #include "avfilter.h"
  #include "filters.h"
  #include "formats.h"
  #include "internal.h"
  #include "af_afir.h"
  
  static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
  {
      int n;
  
      for (n = 0; n < len; n++) {
          const float cre = c[2 * n    ];
          const float cim = c[2 * n + 1];
          const float tre = t[2 * n    ];
          const float tim = t[2 * n + 1];
  
          sum[2 * n    ] += tre * cre - tim * cim;
          sum[2 * n + 1] += tre * cim + tim * cre;
      }
  
      sum[2 * n] += t[2 * n] * c[2 * n];
  }
  
  static void direct(const float *in, const FFTComplex *ir, int len, float *out)
  {
      for (int n = 0; n < len; n++)
          for (int m = 0; m <= n; m++)
              out[n] += ir[m].re * in[n - m];
  }
  
  static void fir_fadd(AudioFIRContext *s, float *dst, const float *src, int nb_samples)
  {
      if ((nb_samples & 15) == 0 && nb_samples >= 16) {
          s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples);
      } else {
          for (int n = 0; n < nb_samples; n++)
              dst[n] += src[n];
      }
  }
  
  static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
  {
      AudioFIRContext *s = ctx->priv;
      const float *in = (const float *)s->in->extended_data[ch] + offset;
      float *block, *buf, *ptr = (float *)out->extended_data[ch] + offset;
      const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset);
      int n, i, j;
  
      for (int segment = 0; segment < s->nb_segments; segment++) {
          AudioFIRSegment *seg = &s->seg[segment];
          float *src = (float *)seg->input->extended_data[ch];
          float *dst = (float *)seg->output->extended_data[ch];
          float *sum = (float *)seg->sum->extended_data[ch];
  
          if (s->min_part_size >= 8) {
              s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4));
              emms_c();
          } else {
              for (n = 0; n < nb_samples; n++)
                  src[seg->input_offset + n] = in[n] * s->dry_gain;
          }
  
          seg->output_offset[ch] += s->min_part_size;
          if (seg->output_offset[ch] == seg->part_size) {
              seg->output_offset[ch] = 0;
          } else {
              memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
  
              dst += seg->output_offset[ch];
              fir_fadd(s, ptr, dst, nb_samples);
              continue;
          }
  
          if (seg->part_size < 8) {
              memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions);
  
              j = seg->part_index[ch];
  
              for (i = 0; i < seg->nb_partitions; i++) {
                  const int coffset = j * seg->coeff_size;
                  const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
  
                  direct(src, coeff, nb_samples, dst);
  
                  if (j == 0)
                      j = seg->nb_partitions;
                  j--;
              }
  
              seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
  
              memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
  
              for (n = 0; n < nb_samples; n++) {
                  ptr[n] += dst[n];
              }
              continue;
          }
  
          memset(sum, 0, sizeof(*sum) * seg->fft_length);
          block = (float *)seg->block->extended_data[ch] + seg->part_index[ch] * seg->block_size;
          memset(block + seg->part_size, 0, sizeof(*block) * (seg->fft_length - seg->part_size));
  
          memcpy(block, src, sizeof(*src) * seg->part_size);
  
          av_rdft_calc(seg->rdft[ch], block);
          block[2 * seg->part_size] = block[1];
          block[1] = 0;
  
          j = seg->part_index[ch];
  
          for (i = 0; i < seg->nb_partitions; i++) {
              const int coffset = j * seg->coeff_size;
              const float *block = (const float *)seg->block->extended_data[ch] + i * seg->block_size;
              const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
  
              s->afirdsp.fcmul_add(sum, block, (const float *)coeff, seg->part_size);
  
              if (j == 0)
                  j = seg->nb_partitions;
              j--;
          }
  
          sum[1] = sum[2 * seg->part_size];
          av_rdft_calc(seg->irdft[ch], sum);
  
          buf = (float *)seg->buffer->extended_data[ch];
          fir_fadd(s, buf, sum, seg->part_size);
  
          memcpy(dst, buf, seg->part_size * sizeof(*dst));
  
          buf = (float *)seg->buffer->extended_data[ch];
          memcpy(buf, sum + seg->part_size, seg->part_size * sizeof(*buf));
  
          seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
  
          memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
  
          fir_fadd(s, ptr, dst, nb_samples);
      }
  
      if (s->min_part_size >= 8) {
          s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4));
          emms_c();
      } else {
          for (n = 0; n < nb_samples; n++)
              ptr[n] *= s->wet_gain;
      }
  
      return 0;
  }
  
  static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
  {
      AudioFIRContext *s = ctx->priv;
  
      for (int offset = 0; offset < out->nb_samples; offset += s->min_part_size) {
          fir_quantum(ctx, out, ch, offset);
      }
  
      return 0;
  }
  
  static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
  {
      AVFrame *out = arg;
      const int start = (out->channels * jobnr) / nb_jobs;
      const int end = (out->channels * (jobnr+1)) / nb_jobs;
  
      for (int ch = start; ch < end; ch++) {
          fir_channel(ctx, out, ch);
      }
  
      return 0;
  }
  
  static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
  {
      AVFilterContext *ctx = outlink->src;
      AVFrame *out = NULL;
  
      out = ff_get_audio_buffer(outlink, in->nb_samples);
      if (!out) {
          av_frame_free(&in);
          return AVERROR(ENOMEM);
      }
  
      if (s->pts == AV_NOPTS_VALUE)
          s->pts = in->pts;
      s->in = in;
      ff_filter_execute(ctx, fir_channels, out, NULL,
                        FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx)));
  
      out->pts = s->pts;
      if (s->pts != AV_NOPTS_VALUE)
          s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
  
      av_frame_free(&in);
      s->in = NULL;
  
      return ff_filter_frame(outlink, out);
  }
  
  static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
  {
      const uint8_t *font;
      int font_height;
      int i;
  
      font = avpriv_cga_font, font_height = 8;
  
      for (i = 0; txt[i]; i++) {
          int char_y, mask;
  
          uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
          for (char_y = 0; char_y < font_height; char_y++) {
              for (mask = 0x80; mask; mask >>= 1) {
                  if (font[txt[i] * font_height + char_y] & mask)
                      AV_WL32(p, color);
                  p += 4;
              }
              p += pic->linesize[0] - 8 * 4;
          }
      }
  }
  
  static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
  {
      int dx = FFABS(x1-x0);
      int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
      int err = (dx>dy ? dx : -dy) / 2, e2;
  
      for (;;) {
          AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
  
          if (x0 == x1 && y0 == y1)
              break;
  
          e2 = err;
  
          if (e2 >-dx) {
              err -= dy;
              x0--;
          }
  
          if (e2 < dy) {
              err += dx;
              y0 += sy;
          }
      }
  }
  
  static void draw_response(AVFilterContext *ctx, AVFrame *out)
  {
      AudioFIRContext *s = ctx->priv;
      float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
      float min_delay = FLT_MAX, max_delay = FLT_MIN;
      int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
      char text[32];
      int channel, i, x;
  
      memset(out->data[0], 0, s->h * out->linesize[0]);
  
      phase = av_malloc_array(s->w, sizeof(*phase));
      mag = av_malloc_array(s->w, sizeof(*mag));
      delay = av_malloc_array(s->w, sizeof(*delay));
      if (!mag || !phase || !delay)
          goto end;
  
      channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->channels - 1);
      for (i = 0; i < s->w; i++) {
          const float *src = (const float *)s->ir[s->selir]->extended_data[channel];
          double w = i * M_PI / (s->w - 1);
          double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
  
          for (x = 0; x < s->nb_taps; x++) {
              real += cos(-x * w) * src[x];
              imag += sin(-x * w) * src[x];
              real_num += cos(-x * w) * src[x] * x;
              imag_num += sin(-x * w) * src[x] * x;
          }
  
          mag[i] = hypot(real, imag);
          phase[i] = atan2(imag, real);
          div = real * real + imag * imag;
          delay[i] = (real_num * real + imag_num * imag) / div;
          min = fminf(min, mag[i]);
          max = fmaxf(max, mag[i]);
          min_delay = fminf(min_delay, delay[i]);
          max_delay = fmaxf(max_delay, delay[i]);
      }
  
      for (i = 0; i < s->w; i++) {
          int ymag = mag[i] / max * (s->h - 1);
          int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
          int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
  
          ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
          yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
          ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
  
          if (prev_ymag < 0)
              prev_ymag = ymag;
          if (prev_yphase < 0)
              prev_yphase = yphase;
          if (prev_ydelay < 0)
              prev_ydelay = ydelay;
  
          draw_line(out, i,   ymag, FFMAX(i - 1, 0),   prev_ymag, 0xFFFF00FF);
          draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
          draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
  
          prev_ymag   = ymag;
          prev_yphase = yphase;
          prev_ydelay = ydelay;
      }
  
      if (s->w > 400 && s->h > 100) {
          drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
          snprintf(text, sizeof(text), "%.2f", max);
          drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
  
          drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
          snprintf(text, sizeof(text), "%.2f", min);
          drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
  
          drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
          snprintf(text, sizeof(text), "%.2f", max_delay);
          drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
  
          drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
          snprintf(text, sizeof(text), "%.2f", min_delay);
          drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
      }
  
  end:
      av_free(delay);
      av_free(phase);
      av_free(mag);
  }
  
  static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
                          int offset, int nb_partitions, int part_size)
  {
      AudioFIRContext *s = ctx->priv;
  
      seg->rdft  = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));
      seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));
      if (!seg->rdft || !seg->irdft)
          return AVERROR(ENOMEM);
  
      seg->fft_length    = part_size * 2 + 1;
      seg->part_size     = part_size;
      seg->block_size    = FFALIGN(seg->fft_length, 32);
      seg->coeff_size    = FFALIGN(seg->part_size + 1, 32);
      seg->nb_partitions = nb_partitions;
      seg->input_size    = offset + s->min_part_size;
      seg->input_offset  = offset;
  
      seg->part_index    = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index));
      seg->output_offset = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->output_offset));
      if (!seg->part_index || !seg->output_offset)
          return AVERROR(ENOMEM);
  
      for (int ch = 0; ch < ctx->inputs[0]->channels && part_size >= 8; ch++) {
          seg->rdft[ch]  = av_rdft_init(av_log2(2 * part_size), DFT_R2C);
          seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R);
          if (!seg->rdft[ch] || !seg->irdft[ch])
              return AVERROR(ENOMEM);
      }
  
      seg->sum    = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
      seg->block  = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
      seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
      seg->coeff  = ff_get_audio_buffer(ctx->inputs[1 + s->selir], seg->nb_partitions * seg->coeff_size * 2);
      seg->input  = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
      seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
      if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff || !seg->input || !seg->output)
          return AVERROR(ENOMEM);
  
      return 0;
  }
  
  static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
  {
      AudioFIRContext *s = ctx->priv;
  
      if (seg->rdft) {
          for (int ch = 0; ch < s->nb_channels; ch++) {
              av_rdft_end(seg->rdft[ch]);
          }
      }
      av_freep(&seg->rdft);
  
      if (seg->irdft) {
          for (int ch = 0; ch < s->nb_channels; ch++) {
              av_rdft_end(seg->irdft[ch]);
          }
      }
      av_freep(&seg->irdft);
  
      av_freep(&seg->output_offset);
      av_freep(&seg->part_index);
  
      av_frame_free(&seg->block);
      av_frame_free(&seg->sum);
      av_frame_free(&seg->buffer);
      av_frame_free(&seg->coeff);
      av_frame_free(&seg->input);
      av_frame_free(&seg->output);
      seg->input_size = 0;
  }
  
  static int convert_coeffs(AVFilterContext *ctx)
  {
      AudioFIRContext *s = ctx->priv;
      int ret, i, ch, n, cur_nb_taps;
      float power = 0;
  
      if (!s->nb_taps) {
          int part_size, max_part_size;
          int left, offset = 0;
  
          s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1 + s->selir]);
          if (s->nb_taps <= 0)
              return AVERROR(EINVAL);
  
          if (s->minp > s->maxp) {
              s->maxp = s->minp;
          }
  
          left = s->nb_taps;
          part_size = 1 << av_log2(s->minp);
          max_part_size = 1 << av_log2(s->maxp);
  
          s->min_part_size = part_size;
  
          for (i = 0; left > 0; i++) {
              int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0);
              int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
  
              s->nb_segments = i + 1;
              ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size);
              if (ret < 0)
                  return ret;
              offset += nb_partitions * part_size;
              left -= nb_partitions * part_size;
              part_size *= 2;
              part_size = FFMIN(part_size, max_part_size);
          }
      }
  
      if (!s->ir[s->selir]) {
          ret = ff_inlink_consume_samples(ctx->inputs[1 + s->selir], s->nb_taps, s->nb_taps, &s->ir[s->selir]);
          if (ret < 0)
              return ret;
          if (ret == 0)
              return AVERROR_BUG;
      }
  
      if (s->response)
          draw_response(ctx, s->video);
  
      s->gain = 1;
      cur_nb_taps = s->ir[s->selir]->nb_samples;
  
      switch (s->gtype) {
      case -1:
          /* nothing to do */
          break;
      case 0:
          for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
              float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
  
              for (i = 0; i < cur_nb_taps; i++)
                  power += FFABS(time[i]);
          }
          s->gain = ctx->inputs[1 + s->selir]->channels / power;
          break;
      case 1:
          for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
              float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
  
              for (i = 0; i < cur_nb_taps; i++)
                  power += time[i];
          }
          s->gain = ctx->inputs[1 + s->selir]->channels / power;
          break;
      case 2:
          for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
              float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
  
              for (i = 0; i < cur_nb_taps; i++)
                  power += time[i] * time[i];
          }
          s->gain = sqrtf(ch / power);
          break;
      default:
          return AVERROR_BUG;
      }
  
      s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
      av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
      for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
          float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
  
          s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 4));
      }
  
      av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", cur_nb_taps);
      av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
  
      for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
          float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
          int toffset = 0;
  
          for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
              time[i] = 0;
  
          av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
  
          for (int segment = 0; segment < s->nb_segments; segment++) {
              AudioFIRSegment *seg = &s->seg[segment];
              float *block = (float *)seg->block->extended_data[ch];
              FFTComplex *coeff = (FFTComplex *)seg->coeff->extended_data[ch];
  
              av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
  
              for (i = 0; i < seg->nb_partitions; i++) {
                  const float scale = 1.f / seg->part_size;
                  const int coffset = i * seg->coeff_size;
                  const int remaining = s->nb_taps - toffset;
                  const int size = remaining >= seg->part_size ? seg->part_size : remaining;
  
                  if (size < 8) {
                      for (n = 0; n < size; n++)
                          coeff[coffset + n].re = time[toffset + n];
  
                      toffset += size;
                      continue;
                  }
  
                  memset(block, 0, sizeof(*block) * seg->fft_length);
                  memcpy(block, time + toffset, size * sizeof(*block));
  
                  av_rdft_calc(seg->rdft[0], block);
  
                  coeff[coffset].re = block[0] * scale;
                  coeff[coffset].im = 0;
                  for (n = 1; n < seg->part_size; n++) {
                      coeff[coffset + n].re = block[2 * n] * scale;
                      coeff[coffset + n].im = block[2 * n + 1] * scale;
                  }
                  coeff[coffset + seg->part_size].re = block[1] * scale;
                  coeff[coffset + seg->part_size].im = 0;
  
                  toffset += size;
              }
  
              av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
              av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
              av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
              av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
              av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
              av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
              av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
          }
      }
  
      s->have_coeffs = 1;
  
      return 0;
  }
  
  static int check_ir(AVFilterLink *link)
  {
      AVFilterContext *ctx = link->dst;
      AudioFIRContext *s = ctx->priv;
      int nb_taps, max_nb_taps;
  
      nb_taps = ff_inlink_queued_samples(link);
      max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
      if (nb_taps > max_nb_taps) {
          av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
          return AVERROR(EINVAL);
      }
  
      return 0;
  }
  
  static int activate(AVFilterContext *ctx)
  {
      AudioFIRContext *s = ctx->priv;
      AVFilterLink *outlink = ctx->outputs[0];
      int ret, status, available, wanted;
      AVFrame *in = NULL;
      int64_t pts;
  
      FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
      if (s->response)
          FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
      if (!s->eof_coeffs[s->selir]) {
          ret = check_ir(ctx->inputs[1 + s->selir]);
          if (ret < 0)
              return ret;
  
          if (ff_outlink_get_status(ctx->inputs[1 + s->selir]) == AVERROR_EOF)
              s->eof_coeffs[s->selir] = 1;
  
          if (!s->eof_coeffs[s->selir]) {
              if (ff_outlink_frame_wanted(ctx->outputs[0]))
                  ff_inlink_request_frame(ctx->inputs[1 + s->selir]);
              else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
                  ff_inlink_request_frame(ctx->inputs[1 + s->selir]);
              return 0;
          }
      }
  
      if (!s->have_coeffs && s->eof_coeffs[s->selir]) {
          ret = convert_coeffs(ctx);
          if (ret < 0)
              return ret;
      }
  
      available = ff_inlink_queued_samples(ctx->inputs[0]);
      wanted = FFMAX(s->min_part_size, (available / s->min_part_size) * s->min_part_size);
      ret = ff_inlink_consume_samples(ctx->inputs[0], wanted, wanted, &in);
      if (ret > 0)
          ret = fir_frame(s, in, outlink);
  
      if (ret < 0)
          return ret;
  
      if (s->response && s->have_coeffs) {
          int64_t old_pts = s->video->pts;
          int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);
  
          if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
              AVFrame *clone;
              s->video->pts = new_pts;
              clone = av_frame_clone(s->video);
              if (!clone)
                  return AVERROR(ENOMEM);
              return ff_filter_frame(ctx->outputs[1], clone);
          }
      }
  
      if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
          ff_filter_set_ready(ctx, 10);
          return 0;
      }
  
      if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
          if (status == AVERROR_EOF) {
              ff_outlink_set_status(ctx->outputs[0], status, pts);
              if (s->response)
                  ff_outlink_set_status(ctx->outputs[1], status, pts);
              return 0;
          }
      }
  
      if (ff_outlink_frame_wanted(ctx->outputs[0]) &&
          !ff_outlink_get_status(ctx->inputs[0])) {
          ff_inlink_request_frame(ctx->inputs[0]);
          return 0;
      }
  
      if (s->response &&
          ff_outlink_frame_wanted(ctx->outputs[1]) &&
          !ff_outlink_get_status(ctx->inputs[0])) {
          ff_inlink_request_frame(ctx->inputs[0]);
          return 0;
      }
  
      return FFERROR_NOT_READY;
  }
  
  static int query_formats(AVFilterContext *ctx)
  {
      AudioFIRContext *s = ctx->priv;
      static const enum AVSampleFormat sample_fmts[] = {
          AV_SAMPLE_FMT_FLTP,
          AV_SAMPLE_FMT_NONE
      };
      static const enum AVPixelFormat pix_fmts[] = {
          AV_PIX_FMT_RGB0,
          AV_PIX_FMT_NONE
      };
      int ret;
  
      if (s->response) {
          AVFilterLink *videolink = ctx->outputs[1];
          AVFilterFormats *formats = ff_make_format_list(pix_fmts);
          if ((ret = ff_formats_ref(formats, &videolink->incfg.formats)) < 0)
              return ret;
      }
  
      if (s->ir_format) {
          ret = ff_set_common_all_channel_counts(ctx);
          if (ret < 0)
              return ret;
      } else {
          AVFilterChannelLayouts *mono = NULL;
          AVFilterChannelLayouts *layouts = ff_all_channel_counts();
  
          if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts)) < 0)
              return ret;
          if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts)) < 0)
              return ret;
  
          ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
          if (ret)
              return ret;
          for (int i = 1; i < ctx->nb_inputs; i++) {
              if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[i]->outcfg.channel_layouts)) < 0)
                  return ret;
          }
      }
  
      if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts)) < 0)
          return ret;
  
      return ff_set_common_all_samplerates(ctx);
  }
  
  static int config_output(AVFilterLink *outlink)
  {
      AVFilterContext *ctx = outlink->src;
      AudioFIRContext *s = ctx->priv;
  
      s->one2many = ctx->inputs[1 + s->selir]->channels == 1;
      outlink->sample_rate = ctx->inputs[0]->sample_rate;
      outlink->time_base   = ctx->inputs[0]->time_base;
      outlink->channel_layout = ctx->inputs[0]->channel_layout;
      outlink->channels = ctx->inputs[0]->channels;
  
      s->nb_channels = outlink->channels;
      s->nb_coef_channels = ctx->inputs[1 + s->selir]->channels;
      s->pts = AV_NOPTS_VALUE;
  
      return 0;
  }
  
  static av_cold void uninit(AVFilterContext *ctx)
  {
      AudioFIRContext *s = ctx->priv;
  
      for (int i = 0; i < s->nb_segments; i++) {
          uninit_segment(ctx, &s->seg[i]);
      }
  
      av_freep(&s->fdsp);
  
      for (int i = 0; i < s->nb_irs; i++) {
          av_frame_free(&s->ir[i]);
      }
  
      av_frame_free(&s->video);
  }
  
  static int config_video(AVFilterLink *outlink)
  {
      AVFilterContext *ctx = outlink->src;
      AudioFIRContext *s = ctx->priv;
  
      outlink->sample_aspect_ratio = (AVRational){1,1};
      outlink->w = s->w;
      outlink->h = s->h;
      outlink->frame_rate = s->frame_rate;
      outlink->time_base = av_inv_q(outlink->frame_rate);
  
      av_frame_free(&s->video);
      s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
      if (!s->video)
          return AVERROR(ENOMEM);
  
      return 0;
  }
  
  void ff_afir_init(AudioFIRDSPContext *dsp)
  {
      dsp->fcmul_add = fcmul_add_c;
  
      if (ARCH_X86)
          ff_afir_init_x86(dsp);
  }
  
  static av_cold int init(AVFilterContext *ctx)
  {
      AudioFIRContext *s = ctx->priv;
      AVFilterPad pad, vpad;
      int ret;
  
      pad = (AVFilterPad) {
          .name = "main",
          .type = AVMEDIA_TYPE_AUDIO,
      };
  
      ret = ff_append_inpad(ctx, &pad);
      if (ret < 0)
          return ret;
  
      for (int n = 0; n < s->nb_irs; n++) {
          pad = (AVFilterPad) {
              .name = av_asprintf("ir%d", n),
              .type = AVMEDIA_TYPE_AUDIO,
          };
  
          if (!pad.name)
              return AVERROR(ENOMEM);
  
          ret = ff_append_inpad_free_name(ctx, &pad);
          if (ret < 0)
              return ret;
      }
  
      pad = (AVFilterPad) {
          .name          = "default",
          .type          = AVMEDIA_TYPE_AUDIO,
          .config_props  = config_output,
      };
  
      ret = ff_append_outpad(ctx, &pad);
      if (ret < 0)
          return ret;
  
      if (s->response) {
          vpad = (AVFilterPad){
              .name         = "filter_response",
              .type         = AVMEDIA_TYPE_VIDEO,
              .config_props = config_video,
          };
  
          ret = ff_append_outpad(ctx, &vpad);
          if (ret < 0)
              return ret;
      }
  
      s->fdsp = avpriv_float_dsp_alloc(0);
      if (!s->fdsp)
          return AVERROR(ENOMEM);
  
      ff_afir_init(&s->afirdsp);
  
      return 0;
  }
  
  static int process_command(AVFilterContext *ctx,
                             const char *cmd,
                             const char *arg,
                             char *res,
                             int res_len,
                             int flags)
  {
      AudioFIRContext *s = ctx->priv;
      int prev_ir = s->selir;
      int ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
  
      if (ret < 0)
          return ret;
  
      s->selir = FFMIN(s->nb_irs - 1, s->selir);
  
      if (prev_ir != s->selir) {
          s->have_coeffs = 0;
      }
  
      return 0;
  }
  
  #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
  #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
  #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
  #define OFFSET(x) offsetof(AudioFIRContext, x)
  
  static const AVOption afir_options[] = {
      { "dry",    "set dry gain",      OFFSET(dry_gain),   AV_OPT_TYPE_FLOAT, {.dbl=1},    0, 10, AF },
      { "wet",    "set wet gain",      OFFSET(wet_gain),   AV_OPT_TYPE_FLOAT, {.dbl=1},    0, 10, AF },
      { "length", "set IR length",     OFFSET(length),     AV_OPT_TYPE_FLOAT, {.dbl=1},    0,  1, AF },
      { "gtype",  "set IR auto gain type",OFFSET(gtype),   AV_OPT_TYPE_INT,   {.i64=0},   -1,  2, AF, "gtype" },
      {  "none",  "without auto gain", 0,                  AV_OPT_TYPE_CONST, {.i64=-1},   0,  0, AF, "gtype" },
      {  "peak",  "peak gain",         0,                  AV_OPT_TYPE_CONST, {.i64=0},    0,  0, AF, "gtype" },
      {  "dc",    "DC gain",           0,                  AV_OPT_TYPE_CONST, {.i64=1},    0,  0, AF, "gtype" },
      {  "gn",    "gain to noise",     0,                  AV_OPT_TYPE_CONST, {.i64=2},    0,  0, AF, "gtype" },
      { "irgain", "set IR gain",       OFFSET(ir_gain),    AV_OPT_TYPE_FLOAT, {.dbl=1},    0,  1, AF },
      { "irfmt",  "set IR format",     OFFSET(ir_format),  AV_OPT_TYPE_INT,   {.i64=1},    0,  1, AF, "irfmt" },
      {  "mono",  "single channel",    0,                  AV_OPT_TYPE_CONST, {.i64=0},    0,  0, AF, "irfmt" },
      {  "input", "same as input",     0,                  AV_OPT_TYPE_CONST, {.i64=1},    0,  0, AF, "irfmt" },
      { "maxir",  "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
      { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
      { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
      { "size",   "set video size",    OFFSET(w),          AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
      { "rate",   "set video rate",    OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
      { "minp",   "set min partition size", OFFSET(minp),  AV_OPT_TYPE_INT,   {.i64=8192}, 1, 32768, AF },
      { "maxp",   "set max partition size", OFFSET(maxp),  AV_OPT_TYPE_INT,   {.i64=8192}, 8, 32768, AF },
      { "nbirs",  "set number of input IRs",OFFSET(nb_irs),AV_OPT_TYPE_INT,   {.i64=1},    1,    32, AF },
      { "ir",     "select IR",              OFFSET(selir), AV_OPT_TYPE_INT,   {.i64=0},    0,    31, AFR },
      { NULL }
  };
  
  AVFILTER_DEFINE_CLASS(afir);
  
  const AVFilter ff_af_afir = {
      .name          = "afir",
      .description   = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in additional stream(s)."),
      .priv_size     = sizeof(AudioFIRContext),
      .priv_class    = &afir_class,
      .query_formats = query_formats,
      .init          = init,
      .activate      = activate,
      .uninit        = uninit,
      .process_command = process_command,
      .flags         = AVFILTER_FLAG_DYNAMIC_INPUTS  |
                       AVFILTER_FLAG_DYNAMIC_OUTPUTS |
                       AVFILTER_FLAG_SLICE_THREADS,
  };
  