2022-05-14 08:28:49 +00:00
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/*
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* Copyright (c) 2017 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avfilter.h"
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#include "formats.h"
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#include "internal.h"
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#include "audio.h"
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#undef ctype
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#undef ftype
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#undef SQRT
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#undef SAMPLE_FORMAT
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#if DEPTH == 32
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#define SAMPLE_FORMAT float
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#define SQRT sqrtf
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#define ctype AVComplexFloat
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#define ftype float
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#else
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#define SAMPLE_FORMAT double
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#define SQRT sqrt
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#define ctype AVComplexDouble
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#define ftype double
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#endif
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#define fn3(a,b) a##_##b
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#define fn2(a,b) fn3(a,b)
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#define fn(a) fn2(a, SAMPLE_FORMAT)
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static void fn(draw_response)(AVFilterContext *ctx, AVFrame *out)
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{
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AudioFIRContext *s = ctx->priv;
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ftype *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
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ftype min_delay = FLT_MAX, max_delay = FLT_MIN;
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int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
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char text[32];
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int channel, i, x;
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memset(out->data[0], 0, s->h * out->linesize[0]);
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phase = av_malloc_array(s->w, sizeof(*phase));
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mag = av_malloc_array(s->w, sizeof(*mag));
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delay = av_malloc_array(s->w, sizeof(*delay));
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if (!mag || !phase || !delay)
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goto end;
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channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->ch_layout.nb_channels - 1);
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for (i = 0; i < s->w; i++) {
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const ftype *src = (const ftype *)s->ir[s->selir]->extended_data[channel];
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double w = i * M_PI / (s->w - 1);
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double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
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for (x = 0; x < s->nb_taps; x++) {
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real += cos(-x * w) * src[x];
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imag += sin(-x * w) * src[x];
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real_num += cos(-x * w) * src[x] * x;
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imag_num += sin(-x * w) * src[x] * x;
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}
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mag[i] = hypot(real, imag);
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phase[i] = atan2(imag, real);
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div = real * real + imag * imag;
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delay[i] = (real_num * real + imag_num * imag) / div;
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min = fminf(min, mag[i]);
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max = fmaxf(max, mag[i]);
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min_delay = fminf(min_delay, delay[i]);
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max_delay = fmaxf(max_delay, delay[i]);
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}
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for (i = 0; i < s->w; i++) {
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int ymag = mag[i] / max * (s->h - 1);
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int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
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int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
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ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
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yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
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ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
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if (prev_ymag < 0)
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prev_ymag = ymag;
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if (prev_yphase < 0)
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prev_yphase = yphase;
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if (prev_ydelay < 0)
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prev_ydelay = ydelay;
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draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
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draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
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draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
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prev_ymag = ymag;
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prev_yphase = yphase;
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prev_ydelay = ydelay;
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}
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if (s->w > 400 && s->h > 100) {
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drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
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snprintf(text, sizeof(text), "%.2f", max);
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drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
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drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
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snprintf(text, sizeof(text), "%.2f", min);
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drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
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drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
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snprintf(text, sizeof(text), "%.2f", max_delay);
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drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
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drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
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snprintf(text, sizeof(text), "%.2f", min_delay);
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drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
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}
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end:
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av_free(delay);
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av_free(phase);
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av_free(mag);
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}
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static void fn(convert_channels)(AVFilterContext *ctx, AudioFIRContext *s)
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{
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for (int ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
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ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
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int toffset = 0;
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for (int i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
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time[i] = 0;
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av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
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for (int segment = 0; segment < s->nb_segments; segment++) {
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AudioFIRSegment *seg = &s->seg[segment];
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ftype *blockin = (ftype *)seg->blockin->extended_data[ch];
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ftype *blockout = (ftype *)seg->blockout->extended_data[ch];
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ctype *coeff = (ctype *)seg->coeff->extended_data[ch];
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av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
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for (int i = 0; i < seg->nb_partitions; i++) {
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const int coffset = i * seg->coeff_size;
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const int remaining = s->nb_taps - toffset;
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const int size = remaining >= seg->part_size ? seg->part_size : remaining;
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if (size < 8) {
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for (int n = 0; n < size; n++)
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coeff[coffset + n].re = time[toffset + n];
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toffset += size;
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continue;
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}
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memset(blockin, 0, sizeof(*blockin) * seg->fft_length);
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memcpy(blockin, time + toffset, size * sizeof(*blockin));
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seg->tx_fn(seg->tx[0], blockout, blockin, sizeof(ftype));
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for (int n = 0; n < seg->part_size + 1; n++) {
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coeff[coffset + n].re = blockout[2 * n];
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coeff[coffset + n].im = blockout[2 * n + 1];
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}
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toffset += size;
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}
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av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
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av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
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av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
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av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
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av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
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av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
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av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
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}
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}
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}
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static int fn(get_power)(AVFilterContext *ctx, AudioFIRContext *s, int cur_nb_taps)
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{
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ftype power = 0;
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int ch;
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switch (s->gtype) {
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case -1:
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/* nothing to do */
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break;
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case 0:
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for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
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ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
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for (int i = 0; i < cur_nb_taps; i++)
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power += FFABS(time[i]);
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}
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s->gain = ctx->inputs[1 + s->selir]->ch_layout.nb_channels / power;
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break;
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case 1:
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for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
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ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
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for (int i = 0; i < cur_nb_taps; i++)
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power += time[i];
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}
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s->gain = ctx->inputs[1 + s->selir]->ch_layout.nb_channels / power;
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break;
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case 2:
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for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
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ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
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for (int i = 0; i < cur_nb_taps; i++)
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power += time[i] * time[i];
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}
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s->gain = SQRT(ch / power);
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break;
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default:
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return AVERROR_BUG;
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}
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s->gain = FFMIN(s->gain * s->ir_gain, 1.);
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av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
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for (int ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
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ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
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#if DEPTH == 32
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s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 4));
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#else
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s->fdsp->vector_dmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 8));
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#endif
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}
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return 0;
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}
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static void fn(direct)(const ftype *in, const ctype *ir, int len, ftype *out)
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{
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for (int n = 0; n < len; n++)
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for (int m = 0; m <= n; m++)
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out[n] += ir[m].re * in[n - m];
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}
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static void fn(fir_fadd)(AudioFIRContext *s, ftype *dst, const ftype *src, int nb_samples)
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{
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if ((nb_samples & 15) == 0 && nb_samples >= 16) {
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#if DEPTH == 32
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s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples);
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#else
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s->fdsp->vector_dmac_scalar(dst, src, 1.0, nb_samples);
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#endif
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} else {
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for (int n = 0; n < nb_samples; n++)
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dst[n] += src[n];
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}
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}
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static int fn(fir_quantum)(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
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{
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AudioFIRContext *s = ctx->priv;
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const ftype *in = (const ftype *)s->in->extended_data[ch] + offset;
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ftype *blockin, *blockout, *buf, *ptr = (ftype *)out->extended_data[ch] + offset;
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const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset);
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int n, i, j;
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for (int segment = 0; segment < s->nb_segments; segment++) {
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AudioFIRSegment *seg = &s->seg[segment];
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ftype *src = (ftype *)seg->input->extended_data[ch];
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ftype *dst = (ftype *)seg->output->extended_data[ch];
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ftype *sumin = (ftype *)seg->sumin->extended_data[ch];
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ftype *sumout = (ftype *)seg->sumout->extended_data[ch];
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if (s->min_part_size >= 8) {
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#if DEPTH == 32
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s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4));
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#else
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s->fdsp->vector_dmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 8));
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#endif
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emms_c();
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} else {
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for (n = 0; n < nb_samples; n++)
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src[seg->input_offset + n] = in[n] * s->dry_gain;
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}
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seg->output_offset[ch] += s->min_part_size;
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if (seg->output_offset[ch] == seg->part_size) {
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seg->output_offset[ch] = 0;
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} else {
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memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
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dst += seg->output_offset[ch];
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fn(fir_fadd)(s, ptr, dst, nb_samples);
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continue;
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}
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if (seg->part_size < 8) {
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memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions);
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j = seg->part_index[ch];
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for (i = 0; i < seg->nb_partitions; i++) {
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const int coffset = j * seg->coeff_size;
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const ctype *coeff = (const ctype *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
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fn(direct)(src, coeff, nb_samples, dst);
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if (j == 0)
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j = seg->nb_partitions;
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j--;
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}
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seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
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memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
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for (n = 0; n < nb_samples; n++) {
|
|
|
|
ptr[n] += dst[n];
|
|
|
|
}
|
|
|
|
continue;
|
|
|
|
}
|
|
|
|
|
|
|
|
memset(sumin, 0, sizeof(*sumin) * seg->fft_length);
|
|
|
|
blockin = (ftype *)seg->blockin->extended_data[ch] + seg->part_index[ch] * seg->block_size;
|
|
|
|
blockout = (ftype *)seg->blockout->extended_data[ch] + seg->part_index[ch] * seg->block_size;
|
|
|
|
memset(blockin + seg->part_size, 0, sizeof(*blockin) * (seg->fft_length - seg->part_size));
|
|
|
|
|
|
|
|
memcpy(blockin, src, sizeof(*src) * seg->part_size);
|
|
|
|
|
|
|
|
seg->tx_fn(seg->tx[ch], blockout, blockin, sizeof(ftype));
|
|
|
|
|
|
|
|
j = seg->part_index[ch];
|
|
|
|
|
|
|
|
for (i = 0; i < seg->nb_partitions; i++) {
|
|
|
|
const int coffset = j * seg->coeff_size;
|
|
|
|
const ftype *blockout = (const ftype *)seg->blockout->extended_data[ch] + i * seg->block_size;
|
|
|
|
const ctype *coeff = (const ctype *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
|
|
|
|
|
|
|
|
#if DEPTH == 32
|
|
|
|
s->afirdsp.fcmul_add(sumin, blockout, (const ftype *)coeff, seg->part_size);
|
|
|
|
#else
|
|
|
|
s->afirdsp.dcmul_add(sumin, blockout, (const ftype *)coeff, seg->part_size);
|
|
|
|
#endif
|
|
|
|
|
|
|
|
if (j == 0)
|
|
|
|
j = seg->nb_partitions;
|
|
|
|
j--;
|
|
|
|
}
|
|
|
|
|
2022-11-18 23:04:06 +00:00
|
|
|
seg->itx_fn(seg->itx[ch], sumout, sumin, sizeof(ctype));
|
2022-05-14 08:28:49 +00:00
|
|
|
|
|
|
|
buf = (ftype *)seg->buffer->extended_data[ch];
|
|
|
|
fn(fir_fadd)(s, buf, sumout, seg->part_size);
|
|
|
|
|
|
|
|
memcpy(dst, buf, seg->part_size * sizeof(*dst));
|
|
|
|
|
|
|
|
buf = (ftype *)seg->buffer->extended_data[ch];
|
|
|
|
memcpy(buf, sumout + 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));
|
|
|
|
|
|
|
|
fn(fir_fadd)(s, ptr, dst, nb_samples);
|
|
|
|
}
|
|
|
|
|
|
|
|
if (s->min_part_size >= 8) {
|
|
|
|
#if DEPTH == 32
|
|
|
|
s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4));
|
|
|
|
#else
|
|
|
|
s->fdsp->vector_dmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 8));
|
|
|
|
#endif
|
|
|
|
emms_c();
|
|
|
|
} else {
|
|
|
|
for (n = 0; n < nb_samples; n++)
|
|
|
|
ptr[n] *= s->wet_gain;
|
|
|
|
}
|
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
|