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ffmpeg/libavfilter/af_afir.c
Andreas Rheinhardt b4f5201967 avfilter: Replace query_formats callback with union of list and callback 
If one looks at the many query_formats callbacks in existence,
one will immediately recognize that there is one type of default
callback for video and a slightly different default callback for
audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);"
for video with a filter-specific pix_fmts list. For audio, it is
the same with a filter-specific sample_fmts list together with
ff_set_common_all_samplerates() and ff_set_common_all_channel_counts().

This commit allows to remove the boilerplate query_formats callbacks
by replacing said callback with a union consisting the old callback
and pointers for pixel and sample format arrays. For the not uncommon
case in which these lists only contain a single entry (besides the
sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also
added to the union to store them directly in the AVFilter,
thereby avoiding a relocation.

The state of said union will be contained in a new, dedicated AVFilter
field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t
in order to create a hole for this new field; this is no problem, as
the maximum of all the nb_inputs is four; for nb_outputs it is only
two).

The state's default value coincides with the earlier default of
query_formats being unset, namely that the filter accepts all formats
(and also sample rates and channel counts/layouts for audio)
provided that these properties agree coincide for all inputs and
outputs.

By using different union members for audio and video filters
the type-unsafety of using the same functions for audio and video
lists will furthermore be more confined to formats.c than before.

When the new fields are used, they will also avoid allocations:
Currently something nearly equivalent to ff_default_query_formats()
is called after every successful call to a query_formats callback;
yet in the common case that the newly allocated AVFilterFormats
are not used at all (namely if there are no free links) these newly
allocated AVFilterFormats are freed again without ever being used.
Filters no longer using the callback will not exhibit this any more.

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-10-05 17:48:25 +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
*/
/**
* @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,
FILTER_QUERY_FUNC(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,
};
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