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FFmpeg/libavfilter/af_afir.c
Andreas Rheinhardt 6d15643173 avfilter/internal: Don't include video.h 
internal.h does not depend on video.h (and should not depend on it)
and therefore should not include video.h at all; instead all users
of video.h should include it directly.

Doing so also avoids unnecessary video.h inclusions in files that
don't need it, like most audio filters.

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
*/
/**
* @file
* An arbitrary audio FIR filter
*/
#include <float.h>
#include "libavutil/cpu.h"
#include "libavutil/tx.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/frame.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/log.h"
#include "libavutil/opt.h"
#include "libavutil/rational.h"
#include "libavutil/xga_font_data.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "formats.h"
#include "internal.h"
#include "af_afir.h"
#include "af_afirdsp.h"
#include "video.h"
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;
}
}
}
#define DEPTH 32
#include "afir_template.c"
#undef DEPTH
#define DEPTH 64
#include "afir_template.c"
static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
{
AudioFIRContext *s = ctx->priv;
const int min_part_size = s->min_part_size;
const int prev_selir = s->prev_selir;
const int selir = s->selir;
for (int offset = 0; offset < out->nb_samples; offset += min_part_size) {
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
if (prev_selir != selir && s->loading[ch] != 0) {
const float *xfade0 = (const float *)s->xfade[0]->extended_data[ch];
const float *xfade1 = (const float *)s->xfade[1]->extended_data[ch];
float *src0 = (float *)s->fadein[0]->extended_data[ch];
float *src1 = (float *)s->fadein[1]->extended_data[ch];
float *dst = ((float *)out->extended_data[ch]) + offset;
memset(src0, 0, min_part_size * sizeof(float));
memset(src1, 0, min_part_size * sizeof(float));
fir_quantum_float(ctx, s->fadein[0], ch, offset, 0, prev_selir);
fir_quantum_float(ctx, s->fadein[1], ch, offset, 0, selir);
if (s->loading[ch] > s->max_offset[selir]) {
for (int n = 0; n < min_part_size; n++)
dst[n] = xfade1[n] * src0[n] + xfade0[n] * src1[n];
s->loading[ch] = 0;
} else {
memcpy(dst, src0, min_part_size * sizeof(float));
}
} else {
fir_quantum_float(ctx, out, ch, offset, offset, selir);
}
break;
case AV_SAMPLE_FMT_DBLP:
if (prev_selir != selir && s->loading[ch] != 0) {
const double *xfade0 = (const double *)s->xfade[0]->extended_data[ch];
const double *xfade1 = (const double *)s->xfade[1]->extended_data[ch];
double *src0 = (double *)s->fadein[0]->extended_data[ch];
double *src1 = (double *)s->fadein[1]->extended_data[ch];
double *dst = ((double *)out->extended_data[ch]) + offset;
memset(src0, 0, min_part_size * sizeof(double));
memset(src1, 0, min_part_size * sizeof(double));
fir_quantum_double(ctx, s->fadein[0], ch, offset, 0, prev_selir);
fir_quantum_double(ctx, s->fadein[1], ch, offset, 0, selir);
if (s->loading[ch] > s->max_offset[selir]) {
for (int n = 0; n < min_part_size; n++)
dst[n] = xfade1[n] * src0[n] + xfade0[n] * src1[n];
s->loading[ch] = 0;
} else {
memcpy(dst, src0, min_part_size * sizeof(double));
}
} else {
fir_quantum_double(ctx, out, ch, offset, offset, selir);
}
break;
}
if (selir != prev_selir && s->loading[ch] != 0)
s->loading[ch] += min_part_size;
}
return 0;
}
static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AVFrame *out = arg;
const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
const int end = (out->ch_layout.nb_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;
out = ff_get_audio_buffer(outlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
out->pts = s->pts = in->pts;
s->in = in;
ff_filter_execute(ctx, fir_channels, out, NULL,
FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
av_frame_free(&in);
s->in = NULL;
return ff_filter_frame(outlink, out);
}
static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int selir,
int offset, int nb_partitions, int part_size, int index)
{
AudioFIRContext *s = ctx->priv;
const size_t cpu_align = av_cpu_max_align();
union { double d; float f; } cscale, scale, iscale;
enum AVTXType tx_type;
int ret;
seg->tx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->tx));
seg->ctx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->ctx));
seg->itx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->itx));
if (!seg->tx || !seg->ctx || !seg->itx)
return AVERROR(ENOMEM);
seg->fft_length = (part_size + 1) * 2;
seg->part_size = part_size;
seg->coeff_size = FFALIGN(seg->part_size + 1, cpu_align);
seg->block_size = FFMAX(seg->coeff_size * 2, FFALIGN(seg->fft_length, cpu_align));
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]->ch_layout.nb_channels, sizeof(*seg->part_index));
seg->output_offset = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->output_offset));
if (!seg->part_index || !seg->output_offset)
return AVERROR(ENOMEM);
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
cscale.f = 1.f;
scale.f = 1.f / sqrtf(2.f * part_size);
iscale.f = 1.f / sqrtf(2.f * part_size);
tx_type = AV_TX_FLOAT_RDFT;
break;
case AV_SAMPLE_FMT_DBLP:
cscale.d = 1.0;
scale.d = 1.0 / sqrt(2.0 * part_size);
iscale.d = 1.0 / sqrt(2.0 * part_size);
tx_type = AV_TX_DOUBLE_RDFT;
break;
}
for (int ch = 0; ch < ctx->inputs[0]->ch_layout.nb_channels && part_size >= 1; ch++) {
ret = av_tx_init(&seg->ctx[ch], &seg->ctx_fn, tx_type,
0, 2 * part_size, &cscale, 0);
if (ret < 0)
return ret;
ret = av_tx_init(&seg->tx[ch], &seg->tx_fn, tx_type,
0, 2 * part_size, &scale, 0);
if (ret < 0)
return ret;
ret = av_tx_init(&seg->itx[ch], &seg->itx_fn, tx_type,
1, 2 * part_size, &iscale, 0);
if (ret < 0)
return ret;
}
seg->sumin = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
seg->sumout = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
seg->blockout = ff_get_audio_buffer(ctx->inputs[0], seg->block_size * seg->nb_partitions);
seg->tempin = ff_get_audio_buffer(ctx->inputs[0], seg->block_size);
seg->tempout = ff_get_audio_buffer(ctx->inputs[0], seg->block_size);
seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size * 5);
if (!seg->buffer || !seg->sumin || !seg->sumout || !seg->blockout ||
!seg->input || !seg->output || !seg->tempin || !seg->tempout)
return AVERROR(ENOMEM);
return 0;
}
static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
{
AudioFIRContext *s = ctx->priv;
if (seg->ctx) {
for (int ch = 0; ch < s->nb_channels; ch++)
av_tx_uninit(&seg->ctx[ch]);
}
av_freep(&seg->ctx);
if (seg->tx) {
for (int ch = 0; ch < s->nb_channels; ch++)
av_tx_uninit(&seg->tx[ch]);
}
av_freep(&seg->tx);
if (seg->itx) {
for (int ch = 0; ch < s->nb_channels; ch++)
av_tx_uninit(&seg->itx[ch]);
}
av_freep(&seg->itx);
av_freep(&seg->output_offset);
av_freep(&seg->part_index);
av_frame_free(&seg->tempin);
av_frame_free(&seg->tempout);
av_frame_free(&seg->blockout);
av_frame_free(&seg->sumin);
av_frame_free(&seg->sumout);
av_frame_free(&seg->buffer);
av_frame_free(&seg->input);
av_frame_free(&seg->output);
seg->input_size = 0;
for (int i = 0; i < MAX_IR_STREAMS; i++)
av_frame_free(&seg->coeff);
}
static int convert_coeffs(AVFilterContext *ctx, int selir)
{
AudioFIRContext *s = ctx->priv;
int ret, nb_taps, cur_nb_taps;
if (!s->nb_taps[selir]) {
int part_size, max_part_size;
int left, offset = 0;
s->nb_taps[selir] = ff_inlink_queued_samples(ctx->inputs[1 + selir]);
if (s->nb_taps[selir] <= 0)
return AVERROR(EINVAL);
if (s->minp > s->maxp)
s->maxp = s->minp;
if (s->nb_segments[selir])
goto skip;
left = s->nb_taps[selir];
part_size = 1 << av_log2(s->minp);
max_part_size = 1 << av_log2(s->maxp);
for (int 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[selir] = i + 1;
ret = init_segment(ctx, &s->seg[selir][i], selir, offset, nb_partitions, part_size, i);
if (ret < 0)
return ret;
offset += nb_partitions * part_size;
s->max_offset[selir] = offset;
left -= nb_partitions * part_size;
part_size *= 2;
part_size = FFMIN(part_size, max_part_size);
}
}
skip:
if (!s->ir[selir]) {
ret = ff_inlink_consume_samples(ctx->inputs[1 + selir], s->nb_taps[selir], s->nb_taps[selir], &s->ir[selir]);
if (ret < 0)
return ret;
if (ret == 0)
return AVERROR_BUG;
}
if (s->response) {
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
draw_response_float(ctx, s->video);
break;
case AV_SAMPLE_FMT_DBLP:
draw_response_double(ctx, s->video);
break;
}
}
cur_nb_taps = s->ir[selir]->nb_samples;
nb_taps = cur_nb_taps;
if (!s->norm_ir[selir] || s->norm_ir[selir]->nb_samples < nb_taps) {
av_frame_free(&s->norm_ir[selir]);
s->norm_ir[selir] = ff_get_audio_buffer(ctx->inputs[0], FFALIGN(nb_taps, 8));
if (!s->norm_ir[selir])
return AVERROR(ENOMEM);
}
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[selir]);
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int ch = 0; ch < s->nb_channels; ch++) {
const float *tsrc = (const float *)s->ir[selir]->extended_data[!s->one2many * ch];
float *time = (float *)s->norm_ir[selir]->extended_data[ch];
memcpy(time, tsrc, sizeof(*time) * nb_taps);
for (int i = FFMAX(1, s->length * nb_taps); i < nb_taps; i++)
time[i] = 0;
get_power_float(ctx, s, nb_taps, ch, time);
for (int n = 0; n < s->nb_segments[selir]; n++) {
AudioFIRSegment *seg = &s->seg[selir][n];
if (!seg->coeff)
seg->coeff = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->coeff_size * 2);
if (!seg->coeff)
return AVERROR(ENOMEM);
for (int i = 0; i < seg->nb_partitions; i++)
convert_channel_float(ctx, s, ch, seg, i, selir);
}
}
break;
case AV_SAMPLE_FMT_DBLP:
for (int ch = 0; ch < s->nb_channels; ch++) {
const double *tsrc = (const double *)s->ir[selir]->extended_data[!s->one2many * ch];
double *time = (double *)s->norm_ir[selir]->extended_data[ch];
memcpy(time, tsrc, sizeof(*time) * nb_taps);
for (int i = FFMAX(1, s->length * nb_taps); i < nb_taps; i++)
time[i] = 0;
get_power_double(ctx, s, nb_taps, ch, time);
for (int n = 0; n < s->nb_segments[selir]; n++) {
AudioFIRSegment *seg = &s->seg[selir][n];
if (!seg->coeff)
seg->coeff = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->coeff_size * 2);
if (!seg->coeff)
return AVERROR(ENOMEM);
for (int i = 0; i < seg->nb_partitions; i++)
convert_channel_double(ctx, s, ch, seg, i, selir);
}
}
break;
}
s->have_coeffs[selir] = 1;
return 0;
}
static int check_ir(AVFilterLink *link, int selir)
{
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);
}
if (ff_inlink_check_available_samples(link, nb_taps + 1) == 1)
s->eof_coeffs[selir] = 1;
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);
for (int i = 0; i < s->nb_irs; i++) {
const int selir = i;
if (s->ir_load && selir != s->selir)
continue;
if (!s->eof_coeffs[selir]) {
ret = check_ir(ctx->inputs[1 + selir], selir);
if (ret < 0)
return ret;
if (!s->eof_coeffs[selir]) {
if (ff_outlink_frame_wanted(ctx->outputs[0]))
ff_inlink_request_frame(ctx->inputs[1 + selir]);
else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
ff_inlink_request_frame(ctx->inputs[1 + selir]);
return 0;
}
}
if (!s->have_coeffs[selir] && s->eof_coeffs[selir]) {
ret = convert_coeffs(ctx, selir);
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 (s->selir != s->prev_selir && s->loading[0] == 0)
s->prev_selir = s->selir;
if (ret < 0)
return ret;
if (s->response && s->have_coeffs[s->selir]) {
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_inlink_request_frame(ctx->inputs[0]);
return 0;
}
if (s->response &&
ff_outlink_frame_wanted(ctx->outputs[1])) {
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[3][3] = {
{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
{ AV_SAMPLE_FMT_DBLP, 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, &(AVChannelLayout)AV_CHANNEL_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[s->precision])) < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioFIRContext *s = ctx->priv;
int ret;
s->one2many = ctx->inputs[1 + s->selir]->ch_layout.nb_channels == 1;
outlink->sample_rate = ctx->inputs[0]->sample_rate;
outlink->time_base = ctx->inputs[0]->time_base;
#if FF_API_OLD_CHANNEL_LAYOUT
FF_DISABLE_DEPRECATION_WARNINGS
outlink->channel_layout = ctx->inputs[0]->channel_layout;
FF_ENABLE_DEPRECATION_WARNINGS
#endif
if ((ret = av_channel_layout_copy(&outlink->ch_layout, &ctx->inputs[0]->ch_layout)) < 0)
return ret;
outlink->ch_layout.nb_channels = ctx->inputs[0]->ch_layout.nb_channels;
s->format = outlink->format;
s->nb_channels = outlink->ch_layout.nb_channels;
s->loading = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->loading));
if (!s->loading)
return AVERROR(ENOMEM);
s->fadein[0] = ff_get_audio_buffer(outlink, s->min_part_size);
s->fadein[1] = ff_get_audio_buffer(outlink, s->min_part_size);
if (!s->fadein[0] || !s->fadein[1])
return AVERROR(ENOMEM);
s->xfade[0] = ff_get_audio_buffer(outlink, s->min_part_size);
s->xfade[1] = ff_get_audio_buffer(outlink, s->min_part_size);
if (!s->xfade[0] || !s->xfade[1])
return AVERROR(ENOMEM);
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int ch = 0; ch < s->nb_channels; ch++) {
float *dst0 = (float *)s->xfade[0]->extended_data[ch];
float *dst1 = (float *)s->xfade[1]->extended_data[ch];
for (int n = 0; n < s->min_part_size; n++) {
dst0[n] = (n + 1.f) / s->min_part_size;
dst1[n] = 1.f - dst0[n];
}
}
break;
case AV_SAMPLE_FMT_DBLP:
for (int ch = 0; ch < s->nb_channels; ch++) {
double *dst0 = (double *)s->xfade[0]->extended_data[ch];
double *dst1 = (double *)s->xfade[1]->extended_data[ch];
for (int n = 0; n < s->min_part_size; n++) {
dst0[n] = (n + 1.0) / s->min_part_size;
dst1[n] = 1.0 - dst0[n];
}
}
break;
}
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
av_freep(&s->fdsp);
av_freep(&s->loading);
for (int i = 0; i < s->nb_irs; i++) {
for (int j = 0; j < s->nb_segments[i]; j++)
uninit_segment(ctx, &s->seg[i][j]);
av_frame_free(&s->ir[i]);
av_frame_free(&s->norm_ir[i]);
}
av_frame_free(&s->fadein[0]);
av_frame_free(&s->fadein[1]);
av_frame_free(&s->xfade[0]);
av_frame_free(&s->xfade[1]);
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;
}
static av_cold int init(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
AVFilterPad pad, vpad;
int ret;
s->prev_selir = FFMIN(s->nb_irs - 1, s->selir);
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);
s->min_part_size = 1 << av_log2(s->minp);
s->max_part_size = 1 << av_log2(s->maxp);
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_selir, ret;
prev_selir = s->selir;
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 (s->selir != prev_selir) {
s->prev_selir = prev_selir;
for (int ch = 0; ch < s->nb_channels; ch++)
s->loading[ch] = 1;
}
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, AFR },
{ "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AFR },
{ "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, 4, 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" },
{ "ac", "AC gain", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "gtype" },
{ "rms", "RMS gain", 0, AV_OPT_TYPE_CONST, {.i64=4}, 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, 65536, AF },
{ "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 65536, 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 },
{ "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, "precision" },
{ "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "precision" },
{ "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision" },
{ "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision" },
{ "irload", "set IR loading type", OFFSET(ir_load), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "irload" },
{ "init", "load all IRs on init", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irload" },
{ "access", "load IR on access", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irload" },
{ 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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