mirror of
https://github.com/FFmpeg/FFmpeg.git
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397 lines
13 KiB
C
397 lines
13 KiB
C
/*
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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 "libavutil/tx.h"
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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 HYPOT
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#undef SAMPLE_FORMAT
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#undef TX_TYPE
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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 HYPOT hypotf
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#define ctype AVComplexFloat
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#define ftype float
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#define TX_TYPE AV_TX_FLOAT_RDFT
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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 HYPOT hypot
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#define ctype AVComplexDouble
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#define ftype double
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#define TX_TYPE AV_TX_DOUBLE_RDFT
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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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for (int y = 0; y < s->h; y++)
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memset(out->data[0] + y * out->linesize[0], 0, s->w * 4);
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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[s->selir]; 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 int fn(get_power)(AVFilterContext *ctx, AudioFIRContext *s,
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int cur_nb_taps, int ch,
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ftype *time)
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{
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ftype ch_gain = 1;
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switch (s->gtype) {
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case -1:
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ch_gain = 1;
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break;
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case 0:
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{
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ftype sum = 0;
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for (int i = 0; i < cur_nb_taps; i++)
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sum += FFABS(time[i]);
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ch_gain = 1. / sum;
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}
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break;
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case 1:
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{
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ftype sum = 0;
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for (int i = 0; i < cur_nb_taps; i++)
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sum += time[i];
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ch_gain = 1. / sum;
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}
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break;
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case 2:
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{
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ftype sum = 0;
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for (int i = 0; i < cur_nb_taps; i++)
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sum += time[i] * time[i];
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ch_gain = 1. / SQRT(sum);
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}
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break;
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case 3:
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case 4:
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{
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ftype *inc, *outc, scale, power;
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AVTXContext *tx;
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av_tx_fn tx_fn;
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int ret, size;
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size = 1 << av_ceil_log2_c(cur_nb_taps);
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inc = av_calloc(size + 2, sizeof(SAMPLE_FORMAT));
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outc = av_calloc(size + 2, sizeof(SAMPLE_FORMAT));
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if (!inc || !outc) {
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av_free(outc);
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av_free(inc);
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break;
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}
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scale = 1.;
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ret = av_tx_init(&tx, &tx_fn, TX_TYPE, 0, size, &scale, 0);
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if (ret < 0) {
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av_free(outc);
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av_free(inc);
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break;
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}
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{
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memcpy(inc, time, cur_nb_taps * sizeof(SAMPLE_FORMAT));
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tx_fn(tx, outc, inc, sizeof(SAMPLE_FORMAT));
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power = 0;
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if (s->gtype == 3) {
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for (int i = 0; i < size / 2 + 1; i++)
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power = FFMAX(power, HYPOT(outc[i * 2], outc[i * 2 + 1]));
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} else {
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ftype sum = 0;
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for (int i = 0; i < size / 2 + 1; i++)
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sum += HYPOT(outc[i * 2], outc[i * 2 + 1]);
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power = SQRT(sum / (size / 2 + 1));
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}
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ch_gain = 1. / power;
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}
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av_tx_uninit(&tx);
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av_free(outc);
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av_free(inc);
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}
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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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if (ch_gain != 1. || s->ir_gain != 1.) {
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ftype gain = ch_gain * s->ir_gain;
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av_log(ctx, AV_LOG_DEBUG, "ch%d gain %f\n", ch, gain);
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#if DEPTH == 32
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s->fdsp->vector_fmul_scalar(time, time, gain, FFALIGN(cur_nb_taps, 4));
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#else
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s->fdsp->vector_dmul_scalar(time, time, 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(convert_channel)(AVFilterContext *ctx, AudioFIRContext *s, int ch,
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AudioFIRSegment *seg, int coeff_partition, int selir)
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{
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const int coffset = coeff_partition * seg->coeff_size;
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const int nb_taps = s->nb_taps[selir];
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ftype *time = (ftype *)s->norm_ir[selir]->extended_data[ch];
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ftype *tempin = (ftype *)seg->tempin->extended_data[ch];
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ftype *tempout = (ftype *)seg->tempout->extended_data[ch];
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ctype *coeff = (ctype *)seg->coeff->extended_data[ch];
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const int remaining = nb_taps - (seg->input_offset + coeff_partition * seg->part_size);
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const int size = remaining >= seg->part_size ? seg->part_size : remaining;
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memset(tempin + size, 0, sizeof(*tempin) * (seg->block_size - size));
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memcpy(tempin, time + seg->input_offset + coeff_partition * seg->part_size,
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size * sizeof(*tempin));
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seg->ctx_fn(seg->ctx[ch], tempout, tempin, sizeof(*tempin));
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memcpy(coeff + coffset, tempout, seg->coeff_size * sizeof(*coeff));
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av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
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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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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 >= 8) {
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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 ioffset, int offset, int selir)
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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] + ioffset;
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ftype *blockout, *ptr = (ftype *)out->extended_data[ch] + offset;
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const int min_part_size = s->min_part_size;
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const int nb_samples = FFMIN(min_part_size, out->nb_samples - offset);
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const int nb_segments = s->nb_segments[selir];
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const float dry_gain = s->dry_gain;
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const float wet_gain = s->wet_gain;
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for (int segment = 0; segment < nb_segments; segment++) {
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AudioFIRSegment *seg = &s->seg[selir][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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ftype *tempin = (ftype *)seg->tempin->extended_data[ch];
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ftype *buf = (ftype *)seg->buffer->extended_data[ch];
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int *output_offset = &seg->output_offset[ch];
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const int nb_partitions = seg->nb_partitions;
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const int input_offset = seg->input_offset;
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const int part_size = seg->part_size;
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int j;
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seg->part_index[ch] = seg->part_index[ch] % nb_partitions;
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if (dry_gain == 1.f) {
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memcpy(src + input_offset, in, nb_samples * sizeof(*src));
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} else if (min_part_size >= 8) {
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#if DEPTH == 32
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s->fdsp->vector_fmul_scalar(src + input_offset, in, dry_gain, FFALIGN(nb_samples, 4));
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#else
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s->fdsp->vector_dmul_scalar(src + input_offset, in, 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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ftype *src2 = src + input_offset;
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for (int n = 0; n < nb_samples; n++)
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src2[n] = in[n] * dry_gain;
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}
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output_offset[0] += min_part_size;
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if (output_offset[0] >= part_size) {
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output_offset[0] = 0;
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} else {
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memmove(src, src + min_part_size, (seg->input_size - min_part_size) * sizeof(*src));
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dst += output_offset[0];
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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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memset(sumin, 0, sizeof(*sumin) * seg->fft_length);
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blockout = (ftype *)seg->blockout->extended_data[ch] + seg->part_index[ch] * seg->block_size;
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memset(tempin + part_size, 0, sizeof(*tempin) * (seg->block_size - part_size));
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memcpy(tempin, src, sizeof(*src) * part_size);
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seg->tx_fn(seg->tx[ch], blockout, tempin, sizeof(ftype));
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j = seg->part_index[ch];
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for (int i = 0; i < nb_partitions; i++) {
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const int input_partition = j;
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const int coeff_partition = i;
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const int coffset = coeff_partition * seg->coeff_size;
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const ftype *blockout = (const ftype *)seg->blockout->extended_data[ch] + input_partition * seg->block_size;
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const ctype *coeff = ((const ctype *)seg->coeff->extended_data[ch]) + coffset;
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if (j == 0)
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j = nb_partitions;
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j--;
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#if DEPTH == 32
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s->afirdsp.fcmul_add(sumin, blockout, (const ftype *)coeff, part_size);
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#else
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s->afirdsp.dcmul_add(sumin, blockout, (const ftype *)coeff, part_size);
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#endif
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}
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seg->itx_fn(seg->itx[ch], sumout, sumin, sizeof(ctype));
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fn(fir_fadd)(s, buf, sumout, part_size);
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memcpy(dst, buf, part_size * sizeof(*dst));
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memcpy(buf, sumout + part_size, part_size * sizeof(*buf));
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fn(fir_fadd)(s, ptr, dst, nb_samples);
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if (part_size != min_part_size)
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memmove(src, src + min_part_size, (seg->input_size - min_part_size) * sizeof(*src));
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seg->part_index[ch] = (seg->part_index[ch] + 1) % nb_partitions;
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}
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if (wet_gain == 1.f)
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return 0;
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if (min_part_size >= 8) {
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#if DEPTH == 32
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s->fdsp->vector_fmul_scalar(ptr, ptr, wet_gain, FFALIGN(nb_samples, 4));
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#else
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s->fdsp->vector_dmul_scalar(ptr, ptr, wet_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 (int n = 0; n < nb_samples; n++)
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ptr[n] *= wet_gain;
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}
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return 0;
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}
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