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FFmpeg/libavfilter/afir_template.c
Andreas Rheinhardt 19ffa2ff2d avfilter: Remove unnecessary formats.h inclusions 
A filter needs formats.h iff it uses FILTER_QUERY_FUNC();
since lots of filters have been switched to use something
else than FILTER_QUERY_FUNC, they don't need it any more,
but removing this header has been forgotten.
This commit does this; files with formats.h inclusion went down
from 304 to 139 here (it were 449 before the preceding commit).

While just at it, also improve the other headers a bit.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2023-08-07 09:21:13 +02:00

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/*
* 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
*/
#include "libavutil/tx.h"
#include "avfilter.h"
#include "internal.h"
#include "audio.h"
#undef ctype
#undef ftype
#undef SQRT
#undef HYPOT
#undef SAMPLE_FORMAT
#undef TX_TYPE
#if DEPTH == 32
#define SAMPLE_FORMAT float
#define SQRT sqrtf
#define HYPOT hypotf
#define ctype AVComplexFloat
#define ftype float
#define TX_TYPE AV_TX_FLOAT_RDFT
#else
#define SAMPLE_FORMAT double
#define SQRT sqrt
#define HYPOT hypot
#define ctype AVComplexDouble
#define ftype double
#define TX_TYPE AV_TX_DOUBLE_RDFT
#endif
#define fn3(a,b) a##_##b
#define fn2(a,b) fn3(a,b)
#define fn(a) fn2(a, SAMPLE_FORMAT)
static void fn(draw_response)(AVFilterContext *ctx, AVFrame *out)
{
AudioFIRContext *s = ctx->priv;
ftype *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
ftype 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;
for (int y = 0; y < s->h; y++)
memset(out->data[0] + y * out->linesize[0], 0, s->w * 4);
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]->ch_layout.nb_channels - 1);
for (i = 0; i < s->w; i++) {
const ftype *src = (const ftype *)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[s->selir]; 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 fn(get_power)(AVFilterContext *ctx, AudioFIRContext *s,
int cur_nb_taps, int ch,
ftype *time)
{
ftype ch_gain = 1;
switch (s->gtype) {
case -1:
ch_gain = 1;
break;
case 0:
{
ftype sum = 0;
for (int i = 0; i < cur_nb_taps; i++)
sum += FFABS(time[i]);
ch_gain = 1. / sum;
}
break;
case 1:
{
ftype sum = 0;
for (int i = 0; i < cur_nb_taps; i++)
sum += time[i];
ch_gain = 1. / sum;
}
break;
case 2:
{
ftype sum = 0;
for (int i = 0; i < cur_nb_taps; i++)
sum += time[i] * time[i];
ch_gain = 1. / SQRT(sum);
}
break;
case 3:
case 4:
{
ftype *inc, *outc, scale, power;
AVTXContext *tx;
av_tx_fn tx_fn;
int ret, size;
size = 1 << av_ceil_log2_c(cur_nb_taps);
inc = av_calloc(size + 2, sizeof(SAMPLE_FORMAT));
outc = av_calloc(size + 2, sizeof(SAMPLE_FORMAT));
if (!inc || !outc) {
av_free(outc);
av_free(inc);
break;
}
scale = 1.;
ret = av_tx_init(&tx, &tx_fn, TX_TYPE, 0, size, &scale, 0);
if (ret < 0) {
av_free(outc);
av_free(inc);
break;
}
{
memcpy(inc, time, cur_nb_taps * sizeof(SAMPLE_FORMAT));
tx_fn(tx, outc, inc, sizeof(SAMPLE_FORMAT));
power = 0;
if (s->gtype == 3) {
for (int i = 0; i < size / 2 + 1; i++)
power = FFMAX(power, HYPOT(outc[i * 2], outc[i * 2 + 1]));
} else {
ftype sum = 0;
for (int i = 0; i < size / 2 + 1; i++)
sum += HYPOT(outc[i * 2], outc[i * 2 + 1]);
power = SQRT(sum / (size / 2 + 1));
}
ch_gain = 1. / power;
}
av_tx_uninit(&tx);
av_free(outc);
av_free(inc);
}
break;
default:
return AVERROR_BUG;
}
if (ch_gain != 1. || s->ir_gain != 1.) {
ftype gain = ch_gain * s->ir_gain;
av_log(ctx, AV_LOG_DEBUG, "ch%d gain %f\n", ch, gain);
#if DEPTH == 32
s->fdsp->vector_fmul_scalar(time, time, gain, FFALIGN(cur_nb_taps, 4));
#else
s->fdsp->vector_dmul_scalar(time, time, gain, FFALIGN(cur_nb_taps, 8));
#endif
}
return 0;
}
static void fn(convert_channel)(AVFilterContext *ctx, AudioFIRContext *s, int ch,
AudioFIRSegment *seg, int coeff_partition, int selir)
{
const int coffset = coeff_partition * seg->coeff_size;
const int nb_taps = s->nb_taps[selir];
ftype *time = (ftype *)s->norm_ir[selir]->extended_data[ch];
ftype *tempin = (ftype *)seg->tempin->extended_data[ch];
ftype *tempout = (ftype *)seg->tempout->extended_data[ch];
ctype *coeff = (ctype *)seg->coeff->extended_data[ch];
const int remaining = nb_taps - (seg->input_offset + coeff_partition * seg->part_size);
const int size = remaining >= seg->part_size ? seg->part_size : remaining;
memset(tempin + size, 0, sizeof(*tempin) * (seg->block_size - size));
memcpy(tempin, time + seg->input_offset + coeff_partition * seg->part_size,
size * sizeof(*tempin));
seg->ctx_fn(seg->ctx[ch], tempout, tempin, sizeof(*tempin));
memcpy(coeff + coffset, tempout, seg->coeff_size * sizeof(*coeff));
av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
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);
}
static void fn(fir_fadd)(AudioFIRContext *s, ftype *dst, const ftype *src, int nb_samples)
{
if ((nb_samples & 15) == 0 && nb_samples >= 8) {
#if DEPTH == 32
s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples);
#else
s->fdsp->vector_dmac_scalar(dst, src, 1.0, nb_samples);
#endif
} else {
for (int n = 0; n < nb_samples; n++)
dst[n] += src[n];
}
}
static int fn(fir_quantum)(AVFilterContext *ctx, AVFrame *out, int ch, int ioffset, int offset, int selir)
{
AudioFIRContext *s = ctx->priv;
const ftype *in = (const ftype *)s->in->extended_data[ch] + ioffset;
ftype *blockout, *ptr = (ftype *)out->extended_data[ch] + offset;
const int min_part_size = s->min_part_size;
const int nb_samples = FFMIN(min_part_size, out->nb_samples - offset);
const int nb_segments = s->nb_segments[selir];
const float dry_gain = s->dry_gain;
const float wet_gain = s->wet_gain;
for (int segment = 0; segment < nb_segments; segment++) {
AudioFIRSegment *seg = &s->seg[selir][segment];
ftype *src = (ftype *)seg->input->extended_data[ch];
ftype *dst = (ftype *)seg->output->extended_data[ch];
ftype *sumin = (ftype *)seg->sumin->extended_data[ch];
ftype *sumout = (ftype *)seg->sumout->extended_data[ch];
ftype *tempin = (ftype *)seg->tempin->extended_data[ch];
ftype *buf = (ftype *)seg->buffer->extended_data[ch];
int *output_offset = &seg->output_offset[ch];
const int nb_partitions = seg->nb_partitions;
const int input_offset = seg->input_offset;
const int part_size = seg->part_size;
int j;
seg->part_index[ch] = seg->part_index[ch] % nb_partitions;
if (dry_gain == 1.f) {
memcpy(src + input_offset, in, nb_samples * sizeof(*src));
} else if (min_part_size >= 8) {
#if DEPTH == 32
s->fdsp->vector_fmul_scalar(src + input_offset, in, dry_gain, FFALIGN(nb_samples, 4));
#else
s->fdsp->vector_dmul_scalar(src + input_offset, in, dry_gain, FFALIGN(nb_samples, 8));
#endif
emms_c();
} else {
ftype *src2 = src + input_offset;
for (int n = 0; n < nb_samples; n++)
src2[n] = in[n] * dry_gain;
}
output_offset[0] += min_part_size;
if (output_offset[0] >= part_size) {
output_offset[0] = 0;
} else {
memmove(src, src + min_part_size, (seg->input_size - min_part_size) * sizeof(*src));
dst += output_offset[0];
fn(fir_fadd)(s, ptr, dst, nb_samples);
continue;
}
memset(sumin, 0, sizeof(*sumin) * seg->fft_length);
blockout = (ftype *)seg->blockout->extended_data[ch] + seg->part_index[ch] * seg->block_size;
memset(tempin + part_size, 0, sizeof(*tempin) * (seg->block_size - part_size));
memcpy(tempin, src, sizeof(*src) * part_size);
seg->tx_fn(seg->tx[ch], blockout, tempin, sizeof(ftype));
j = seg->part_index[ch];
for (int i = 0; i < nb_partitions; i++) {
const int input_partition = j;
const int coeff_partition = i;
const int coffset = coeff_partition * seg->coeff_size;
const ftype *blockout = (const ftype *)seg->blockout->extended_data[ch] + input_partition * seg->block_size;
const ctype *coeff = ((const ctype *)seg->coeff->extended_data[ch]) + coffset;
if (j == 0)
j = nb_partitions;
j--;
#if DEPTH == 32
s->afirdsp.fcmul_add(sumin, blockout, (const ftype *)coeff, part_size);
#else
s->afirdsp.dcmul_add(sumin, blockout, (const ftype *)coeff, part_size);
#endif
}
seg->itx_fn(seg->itx[ch], sumout, sumin, sizeof(ctype));
fn(fir_fadd)(s, buf, sumout, part_size);
memcpy(dst, buf, part_size * sizeof(*dst));
memcpy(buf, sumout + part_size, part_size * sizeof(*buf));
fn(fir_fadd)(s, ptr, dst, nb_samples);
if (part_size != min_part_size)
memmove(src, src + min_part_size, (seg->input_size - min_part_size) * sizeof(*src));
seg->part_index[ch] = (seg->part_index[ch] + 1) % nb_partitions;
}
if (wet_gain == 1.f)
return 0;
if (min_part_size >= 8) {
#if DEPTH == 32
s->fdsp->vector_fmul_scalar(ptr, ptr, wet_gain, FFALIGN(nb_samples, 4));
#else
s->fdsp->vector_dmul_scalar(ptr, ptr, wet_gain, FFALIGN(nb_samples, 8));
#endif
emms_c();
} else {
for (int n = 0; n < nb_samples; n++)
ptr[n] *= wet_gain;
}
return 0;
}
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