ffmpeg/libavfilter/af_crossfeed.c

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/*
* 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/mem.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,
2021-08-12 13:05:31 +02:00
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
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>
2021-09-27 12:07:35 +02:00
FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
.process_command = process_command,
};