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FFmpeg/libavfilter/af_afir.c
Andreas Rheinhardt 790f793844 avutil/common: Don't auto-include mem.h 
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.

Keep it for external users in order to not cause breakages.

Also improve the other headers a bit while just at it.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-31 00:08:43 +01:00

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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/mem.h"
#include "libavutil/tx.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavutil/frame.h"
#include "libavutil/log.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "formats.h"
#include "internal.h"
#include "af_afir.h"
#include "af_afirdsp.h"
#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:
fir_quantums_float(ctx, s, out, min_part_size, ch, offset, prev_selir, selir);
break;
case AV_SAMPLE_FMT_DBLP:
fir_quantums_double(ctx, s, out, min_part_size, ch, offset, prev_selir, 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)));
s->prev_is_disabled = ctx->is_disabled;
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;
}
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];
s->ch_gain[ch] = ir_gain_float(ctx, s, nb_taps, tsrc);
}
if (s->ir_link) {
float gain = +INFINITY;
for (int ch = 0; ch < s->nb_channels; ch++)
gain = fminf(gain, s->ch_gain[ch]);
for (int ch = 0; ch < s->nb_channels; ch++)
s->ch_gain[ch] = gain;
}
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;
ir_scale_float(ctx, s, nb_taps, ch, time, s->ch_gain[ch]);
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];
s->ch_gain[ch] = ir_gain_double(ctx, s, nb_taps, tsrc);
}
if (s->ir_link) {
double gain = +INFINITY;
for (int ch = 0; ch < s->nb_channels; ch++)
gain = fmin(gain, s->ch_gain[ch]);
for (int ch = 0; ch < s->nb_channels; ch++)
s->ch_gain[ch] = gain;
}
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;
ir_scale_double(ctx, s, nb_taps, ch, time, s->ch_gain[ch]);
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);
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]);
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 (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);
return 0;
}
}
if (ff_outlink_frame_wanted(ctx->outputs[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[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 },
};
int 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 ((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->ch_gain = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->ch_gain));
s->loading = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->loading));
if (!s->loading || !s->ch_gain)
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->ch_gain);
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]);
}
static av_cold int init(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
AVFilterPad pad;
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;
}
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|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
{ "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
{ "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
{ "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
{ "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
{ "ac", "AC gain", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
{ "rms", "RMS gain", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" },
{ "irnorm", "set IR norm", OFFSET(ir_norm), AV_OPT_TYPE_FLOAT, {.dbl=1}, -1, 2, AF },
{ "irlink", "set IR link", OFFSET(ir_link), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
{ "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, .unit = "irfmt" },
{ "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "irfmt" },
{ "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "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|AV_OPT_FLAG_DEPRECATED },
{ "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF|AV_OPT_FLAG_DEPRECATED },
{ "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF|AV_OPT_FLAG_DEPRECATED },
{ "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF|AV_OPT_FLAG_DEPRECATED },
{ "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, .unit = "precision" },
{ "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" },
{ "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" },
{ "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" },
{ "irload", "set IR loading type", OFFSET(ir_load), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, .unit = "irload" },
{ "init", "load all IRs on init", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "irload" },
{ "access", "load IR on access", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "irload" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(afir);
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
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),
FILTER_OUTPUTS(outputs),
.init = init,
.activate = activate,
.uninit = uninit,
.process_command = process_command,
.flags = AVFILTER_FLAG_DYNAMIC_INPUTS |
AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
};
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