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FFmpeg/libavfilter/af_crossfeed.c
Andreas Rheinhardt 50ea7389ec avfilter: Deduplicate default audio inputs/outputs
Lots of audio filters use very simple inputs or outputs:
An array with a single AVFilterPad whose name is "default"
and whose type is AVMEDIA_TYPE_AUDIO; everything else is unset.

Given that we never use pointer equality for inputs or outputs*,
we can simply use a single AVFilterPad instead of dozens; this
even saves .data.rel.ro (4784B here) as well as relocations.

*: In fact, several filters (like the filters in af_biquads.c)
already use the same inputs; furthermore, ff_filter_alloc()
duplicates the input and output pads so that we do not even
work with the pads directly.

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

378 lines
11 KiB
C

/*
* 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/channel_layout.h"
#include "libavutil/ffmath.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "filters.h"
#include "formats.h"
typedef struct CrossfeedContext {
const AVClass *class;
double range;
double strength;
double slope;
double level_in;
double level_out;
int block_samples;
int block_size;
double a0, a1, a2;
double b0, b1, b2;
double w1, w2;
int64_t pts;
int nb_samples;
double *mid;
double *side[3];
} CrossfeedContext;
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layout = NULL;
int ret;
if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
(ret = ff_set_common_formats (ctx , formats )) < 0 ||
(ret = ff_add_channel_layout (&layout , &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO)) < 0 ||
(ret = ff_set_common_channel_layouts (ctx , layout )) < 0 ||
(ret = ff_set_common_all_samplerates (ctx )) < 0)
return ret;
return 0;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
CrossfeedContext *s = ctx->priv;
double A = ff_exp10(s->strength * -30 / 40);
double w0 = 2 * M_PI * (1. - s->range) * 2100 / inlink->sample_rate;
double alpha;
alpha = sin(w0) / 2 * sqrt((A + 1 / A) * (1 / s->slope - 1) + 2);
s->a0 = (A + 1) + (A - 1) * cos(w0) + 2 * sqrt(A) * alpha;
s->a1 = -2 * ((A - 1) + (A + 1) * cos(w0));
s->a2 = (A + 1) + (A - 1) * cos(w0) - 2 * sqrt(A) * alpha;
s->b0 = A * ((A + 1) - (A - 1) * cos(w0) + 2 * sqrt(A) * alpha);
s->b1 = 2 * A * ((A - 1) - (A + 1) * cos(w0));
s->b2 = A * ((A + 1) - (A - 1) * cos(w0) - 2 * sqrt(A) * alpha);
s->a1 /= s->a0;
s->a2 /= s->a0;
s->b0 /= s->a0;
s->b1 /= s->a0;
s->b2 /= s->a0;
if (s->block_samples == 0 && s->block_size > 0) {
s->block_samples = s->block_size;
s->mid = av_calloc(s->block_samples * 2, sizeof(*s->mid));
for (int i = 0; i < 3; i++) {
s->side[i] = av_calloc(s->block_samples * 2, sizeof(*s->side[0]));
if (!s->side[i])
return AVERROR(ENOMEM);
}
}
return 0;
}
static void reverse_samples(double *dst, const double *src,
int nb_samples)
{
for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--)
dst[i] = src[j];
}
static void filter_samples(double *dst, const double *src,
int nb_samples,
double b0, double b1, double b2,
double a1, double a2,
double *sw1, double *sw2)
{
double w1 = *sw1;
double w2 = *sw2;
for (int n = 0; n < nb_samples; n++) {
double side = src[n];
double oside = side * b0 + w1;
w1 = b1 * side + w2 + a1 * oside;
w2 = b2 * side + a2 * oside;
dst[n] = oside;
}
*sw1 = w1;
*sw2 = w2;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in, int eof)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
CrossfeedContext *s = ctx->priv;
const double *src = (const double *)in->data[0];
const double level_in = s->level_in;
const double level_out = s->level_out;
const double b0 = s->b0;
const double b1 = s->b1;
const double b2 = s->b2;
const double a1 = -s->a1;
const double a2 = -s->a2;
AVFrame *out;
int drop = 0;
double *dst;
if (av_frame_is_writable(in) && s->block_samples == 0) {
out = in;
} else {
out = ff_get_audio_buffer(outlink, s->block_samples > 0 ? s->block_samples : in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
}
dst = (double *)out->data[0];
if (s->block_samples > 0 && s->pts == AV_NOPTS_VALUE)
drop = 1;
if (s->block_samples == 0) {
double w1 = s->w1;
double w2 = s->w2;
for (int n = 0; n < out->nb_samples; n++, src += 2, dst += 2) {
double mid = (src[0] + src[1]) * level_in * .5;
double side = (src[0] - src[1]) * level_in * .5;
double oside = side * b0 + w1;
w1 = b1 * side + w2 + a1 * oside;
w2 = b2 * side + a2 * oside;
if (ctx->is_disabled) {
dst[0] = src[0];
dst[1] = src[1];
} else {
dst[0] = (mid + oside) * level_out;
dst[1] = (mid - oside) * level_out;
}
}
s->w1 = w1;
s->w2 = w2;
} else if (eof) {
const double *src = (const double *)in->data[0];
double *ssrc = s->side[1] + s->block_samples;
double *msrc = s->mid;
for (int n = 0; n < out->nb_samples; n++, src += 2, dst += 2) {
if (ctx->is_disabled) {
dst[0] = src[0];
dst[1] = src[1];
} else {
dst[0] = (msrc[n] + ssrc[n]) * level_out;
dst[1] = (msrc[n] - ssrc[n]) * level_out;
}
}
} else {
double *mdst = s->mid + s->block_samples;
double *sdst = s->side[0] + s->block_samples;
double *ssrc = s->side[0];
double *msrc = s->mid;
double w1 = s->w1;
double w2 = s->w2;
for (int n = 0; n < out->nb_samples; n++, src += 2) {
mdst[n] = (src[0] + src[1]) * level_in * .5;
sdst[n] = (src[0] - src[1]) * level_in * .5;
}
sdst = s->side[1];
filter_samples(sdst, ssrc, s->block_samples,
b0, b1, b2, a1, a2,
&w1, &w2);
s->w1 = w1;
s->w2 = w2;
ssrc = s->side[0] + s->block_samples;
sdst = s->side[1] + s->block_samples;
filter_samples(sdst, ssrc, s->block_samples,
b0, b1, b2, a1, a2,
&w1, &w2);
reverse_samples(s->side[2], s->side[1], s->block_samples * 2);
w1 = w2 = 0.;
filter_samples(s->side[2], s->side[2], s->block_samples * 2,
b0, b1, b2, a1, a2,
&w1, &w2);
reverse_samples(s->side[1], s->side[2], s->block_samples * 2);
src = (const double *)in->data[0];
ssrc = s->side[1];
for (int n = 0; n < out->nb_samples; n++, src += 2, dst += 2) {
if (ctx->is_disabled) {
dst[0] = src[0];
dst[1] = src[1];
} else {
dst[0] = (msrc[n] + ssrc[n]) * level_out;
dst[1] = (msrc[n] - ssrc[n]) * level_out;
}
}
memmove(s->mid, s->mid + s->block_samples,
s->block_samples * sizeof(*s->mid));
memmove(s->side[0], s->side[0] + s->block_samples,
s->block_samples * sizeof(*s->side[0]));
}
if (s->block_samples > 0) {
int nb_samples = in->nb_samples;
int64_t pts = in->pts;
out->pts = s->pts;
out->nb_samples = s->nb_samples;
s->pts = pts;
s->nb_samples = nb_samples;
}
if (out != in)
av_frame_free(&in);
if (!drop) {
return ff_filter_frame(outlink, out);
} else {
av_frame_free(&out);
ff_filter_set_ready(ctx, 10);
return 0;
}
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
CrossfeedContext *s = ctx->priv;
AVFrame *in = NULL;
int64_t pts;
int status;
int ret;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
if (s->block_samples > 0) {
ret = ff_inlink_consume_samples(inlink, s->block_samples, s->block_samples, &in);
} else {
ret = ff_inlink_consume_frame(inlink, &in);
}
if (ret < 0)
return ret;
if (ret > 0)
return filter_frame(inlink, in, 0);
if (s->block_samples > 0 && ff_inlink_queued_samples(inlink) >= s->block_samples) {
ff_filter_set_ready(ctx, 10);
return 0;
}
if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
if (s->block_samples > 0) {
AVFrame *in = ff_get_audio_buffer(outlink, s->block_samples);
if (!in)
return AVERROR(ENOMEM);
ret = filter_frame(inlink, in, 1);
}
ff_outlink_set_status(outlink, status, pts);
return ret;
}
FF_FILTER_FORWARD_WANTED(outlink, inlink);
return FFERROR_NOT_READY;
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
int ret;
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
if (ret < 0)
return ret;
return config_input(ctx->inputs[0]);
}
static av_cold void uninit(AVFilterContext *ctx)
{
CrossfeedContext *s = ctx->priv;
av_freep(&s->mid);
for (int i = 0; i < 3; i++)
av_freep(&s->side[i]);
}
#define OFFSET(x) offsetof(CrossfeedContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption crossfeed_options[] = {
{ "strength", "set crossfeed strength", OFFSET(strength), AV_OPT_TYPE_DOUBLE, {.dbl=.2}, 0, 1, FLAGS },
{ "range", "set soundstage wideness", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, FLAGS },
{ "slope", "set curve slope", OFFSET(slope), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .01, 1, FLAGS },
{ "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=.9}, 0, 1, FLAGS },
{ "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1.}, 0, 1, FLAGS },
{ "block_size", "set the block size", OFFSET(block_size),AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF },
{ NULL }
};
AVFILTER_DEFINE_CLASS(crossfeed);
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
},
};
const AVFilter ff_af_crossfeed = {
.name = "crossfeed",
.description = NULL_IF_CONFIG_SMALL("Apply headphone crossfeed filter."),
.priv_size = sizeof(CrossfeedContext),
.priv_class = &crossfeed_class,
.activate = activate,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
.process_command = process_command,
};