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FFmpeg/libavfilter/avf_ahistogram.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

436 lines
15 KiB
C

/*
* Copyright (c) 2015 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
*/
#include "libavutil/avassert.h"
#include "libavutil/opt.h"
#include "libavutil/parseutils.h"
#include "avfilter.h"
#include "filters.h"
#include "formats.h"
#include "audio.h"
#include "video.h"
#include "internal.h"
enum DisplayScale { LINEAR, SQRT, CBRT, LOG, RLOG, NB_SCALES };
enum AmplitudeScale { ALINEAR, ALOG, NB_ASCALES };
enum SlideMode { REPLACE, SCROLL, NB_SLIDES };
enum DisplayMode { SINGLE, SEPARATE, NB_DMODES };
enum HistogramMode { ACCUMULATE, CURRENT, NB_HMODES };
typedef struct AudioHistogramContext {
const AVClass *class;
AVFrame *out;
int w, h;
AVRational frame_rate;
uint64_t *achistogram;
uint64_t *shistogram;
int ascale;
int scale;
float phisto;
int histogram_h;
int apos;
int ypos;
int slide;
int dmode;
int dchannels;
int count;
int frame_count;
float *combine_buffer;
AVFrame *in[101];
int first;
int nb_samples;
} AudioHistogramContext;
#define OFFSET(x) offsetof(AudioHistogramContext, x)
#define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_VIDEO_PARAM
static const AVOption ahistogram_options[] = {
{ "dmode", "set method to display channels", OFFSET(dmode), AV_OPT_TYPE_INT, {.i64=SINGLE}, 0, NB_DMODES-1, FLAGS, "dmode" },
{ "single", "all channels use single histogram", 0, AV_OPT_TYPE_CONST, {.i64=SINGLE}, 0, 0, FLAGS, "dmode" },
{ "separate", "each channel have own histogram", 0, AV_OPT_TYPE_CONST, {.i64=SEPARATE}, 0, 0, FLAGS, "dmode" },
{ "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str="25"}, 0, INT_MAX, FLAGS },
{ "r", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str="25"}, 0, INT_MAX, FLAGS },
{ "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str="hd720"}, 0, 0, FLAGS },
{ "s", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str="hd720"}, 0, 0, FLAGS },
{ "scale", "set display scale", OFFSET(scale), AV_OPT_TYPE_INT, {.i64=LOG}, LINEAR, NB_SCALES-1, FLAGS, "scale" },
{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=LOG}, 0, 0, FLAGS, "scale" },
{ "sqrt", "square root", 0, AV_OPT_TYPE_CONST, {.i64=SQRT}, 0, 0, FLAGS, "scale" },
{ "cbrt", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64=CBRT}, 0, 0, FLAGS, "scale" },
{ "lin", "linear", 0, AV_OPT_TYPE_CONST, {.i64=LINEAR}, 0, 0, FLAGS, "scale" },
{ "rlog", "reverse logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=RLOG}, 0, 0, FLAGS, "scale" },
{ "ascale", "set amplitude scale", OFFSET(ascale), AV_OPT_TYPE_INT, {.i64=ALOG}, LINEAR, NB_ASCALES-1, FLAGS, "ascale" },
{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=ALOG}, 0, 0, FLAGS, "ascale" },
{ "lin", "linear", 0, AV_OPT_TYPE_CONST, {.i64=ALINEAR}, 0, 0, FLAGS, "ascale" },
{ "acount", "how much frames to accumulate", OFFSET(count), AV_OPT_TYPE_INT, {.i64=1}, -1, 100, FLAGS },
{ "rheight", "set histogram ratio of window height", OFFSET(phisto), AV_OPT_TYPE_FLOAT, {.dbl=0.10}, 0, 1, FLAGS },
{ "slide", "set sonogram sliding", OFFSET(slide), AV_OPT_TYPE_INT, {.i64=REPLACE}, 0, NB_SLIDES-1, FLAGS, "slide" },
{ "replace", "replace old rows with new", 0, AV_OPT_TYPE_CONST, {.i64=REPLACE}, 0, 0, FLAGS, "slide" },
{ "scroll", "scroll from top to bottom", 0, AV_OPT_TYPE_CONST, {.i64=SCROLL}, 0, 0, FLAGS, "slide" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(ahistogram);
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layouts = NULL;
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE };
static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_YUVA444P, AV_PIX_FMT_NONE };
int ret = AVERROR(EINVAL);
formats = ff_make_format_list(sample_fmts);
if ((ret = ff_formats_ref (formats, &inlink->outcfg.formats )) < 0 ||
(layouts = ff_all_channel_counts()) == NULL ||
(ret = ff_channel_layouts_ref (layouts, &inlink->outcfg.channel_layouts)) < 0)
return ret;
formats = ff_all_samplerates();
if ((ret = ff_formats_ref(formats, &inlink->outcfg.samplerates)) < 0)
return ret;
formats = ff_make_format_list(pix_fmts);
if ((ret = ff_formats_ref(formats, &outlink->incfg.formats)) < 0)
return ret;
return 0;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioHistogramContext *s = ctx->priv;
s->nb_samples = FFMAX(1, av_rescale(inlink->sample_rate, s->frame_rate.den, s->frame_rate.num));
s->dchannels = s->dmode == SINGLE ? 1 : inlink->channels;
s->shistogram = av_calloc(s->w, s->dchannels * sizeof(*s->shistogram));
if (!s->shistogram)
return AVERROR(ENOMEM);
s->achistogram = av_calloc(s->w, s->dchannels * sizeof(*s->achistogram));
if (!s->achistogram)
return AVERROR(ENOMEM);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AudioHistogramContext *s = outlink->src->priv;
outlink->w = s->w;
outlink->h = s->h;
outlink->sample_aspect_ratio = (AVRational){1,1};
outlink->frame_rate = s->frame_rate;
s->histogram_h = s->h * s->phisto;
s->ypos = s->h * s->phisto;
if (s->dmode == SEPARATE) {
s->combine_buffer = av_malloc_array(outlink->w * 3, sizeof(*s->combine_buffer));
if (!s->combine_buffer)
return AVERROR(ENOMEM);
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioHistogramContext *s = ctx->priv;
const int H = s->histogram_h;
const int w = s->w;
int c, y, n, p, bin;
uint64_t acmax = 1;
AVFrame *clone;
if (!s->out || s->out->width != outlink->w ||
s->out->height != outlink->h) {
av_frame_free(&s->out);
s->out = ff_get_video_buffer(outlink, outlink->w, outlink->h);
if (!s->out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
for (n = H; n < s->h; n++) {
memset(s->out->data[0] + n * s->out->linesize[0], 0, w);
memset(s->out->data[1] + n * s->out->linesize[0], 127, w);
memset(s->out->data[2] + n * s->out->linesize[0], 127, w);
memset(s->out->data[3] + n * s->out->linesize[0], 0, w);
}
}
if (s->dmode == SEPARATE) {
for (y = 0; y < w; y++) {
s->combine_buffer[3 * y ] = 0;
s->combine_buffer[3 * y + 1] = 127.5;
s->combine_buffer[3 * y + 2] = 127.5;
}
}
for (n = 0; n < H; n++) {
memset(s->out->data[0] + n * s->out->linesize[0], 0, w);
memset(s->out->data[1] + n * s->out->linesize[0], 127, w);
memset(s->out->data[2] + n * s->out->linesize[0], 127, w);
memset(s->out->data[3] + n * s->out->linesize[0], 0, w);
}
s->out->pts = in->pts;
s->first = s->frame_count;
switch (s->ascale) {
case ALINEAR:
for (c = 0; c < inlink->channels; c++) {
const float *src = (const float *)in->extended_data[c];
uint64_t *achistogram = &s->achistogram[(s->dmode == SINGLE ? 0: c) * w];
for (n = 0; n < in->nb_samples; n++) {
bin = lrint(av_clipf(fabsf(src[n]), 0, 1) * (w - 1));
achistogram[bin]++;
}
if (s->in[s->first] && s->count >= 0) {
uint64_t *shistogram = &s->shistogram[(s->dmode == SINGLE ? 0: c) * w];
const float *src2 = (const float *)s->in[s->first]->extended_data[c];
for (n = 0; n < in->nb_samples; n++) {
bin = lrint(av_clipf(fabsf(src2[n]), 0, 1) * (w - 1));
shistogram[bin]++;
}
}
}
break;
case ALOG:
for (c = 0; c < inlink->channels; c++) {
const float *src = (const float *)in->extended_data[c];
uint64_t *achistogram = &s->achistogram[(s->dmode == SINGLE ? 0: c) * w];
for (n = 0; n < in->nb_samples; n++) {
bin = lrint(av_clipf(1 + log10(fabsf(src[n])) / 6, 0, 1) * (w - 1));
achistogram[bin]++;
}
if (s->in[s->first] && s->count >= 0) {
uint64_t *shistogram = &s->shistogram[(s->dmode == SINGLE ? 0: c) * w];
const float *src2 = (const float *)s->in[s->first]->extended_data[c];
for (n = 0; n < in->nb_samples; n++) {
bin = lrint(av_clipf(1 + log10(fabsf(src2[n])) / 6, 0, 1) * (w - 1));
shistogram[bin]++;
}
}
}
break;
}
av_frame_free(&s->in[s->frame_count]);
s->in[s->frame_count] = in;
s->frame_count++;
if (s->frame_count > s->count)
s->frame_count = 0;
for (n = 0; n < w * s->dchannels; n++) {
acmax = FFMAX(s->achistogram[n] - s->shistogram[n], acmax);
}
for (c = 0; c < s->dchannels; c++) {
uint64_t *shistogram = &s->shistogram[c * w];
uint64_t *achistogram = &s->achistogram[c * w];
float yf, uf, vf;
if (s->dmode == SEPARATE) {
yf = 256.0f / s->dchannels;
uf = yf * M_PI;
vf = yf * M_PI;
uf *= 0.5 * sin((2 * M_PI * c) / s->dchannels);
vf *= 0.5 * cos((2 * M_PI * c) / s->dchannels);
}
for (n = 0; n < w; n++) {
double a, aa;
int h;
a = achistogram[n] - shistogram[n];
switch (s->scale) {
case LINEAR:
aa = a / (double)acmax;
break;
case SQRT:
aa = sqrt(a) / sqrt(acmax);
break;
case CBRT:
aa = cbrt(a) / cbrt(acmax);
break;
case LOG:
aa = log2(a + 1) / log2(acmax + 1);
break;
case RLOG:
aa = 1. - log2(a + 1) / log2(acmax + 1);
if (aa == 1.)
aa = 0;
break;
default:
av_assert0(0);
}
h = aa * (H - 1);
if (s->dmode == SINGLE) {
for (y = H - h; y < H; y++) {
s->out->data[0][y * s->out->linesize[0] + n] = 255;
s->out->data[3][y * s->out->linesize[0] + n] = 255;
}
if (s->h - H > 0) {
h = aa * 255;
s->out->data[0][s->ypos * s->out->linesize[0] + n] = h;
s->out->data[1][s->ypos * s->out->linesize[1] + n] = 127;
s->out->data[2][s->ypos * s->out->linesize[2] + n] = 127;
s->out->data[3][s->ypos * s->out->linesize[3] + n] = 255;
}
} else if (s->dmode == SEPARATE) {
float *out = &s->combine_buffer[3 * n];
int old;
old = s->out->data[0][(H - h) * s->out->linesize[0] + n];
for (y = H - h; y < H; y++) {
if (s->out->data[0][y * s->out->linesize[0] + n] != old)
break;
old = s->out->data[0][y * s->out->linesize[0] + n];
s->out->data[0][y * s->out->linesize[0] + n] = yf;
s->out->data[1][y * s->out->linesize[1] + n] = 128+uf;
s->out->data[2][y * s->out->linesize[2] + n] = 128+vf;
s->out->data[3][y * s->out->linesize[3] + n] = 255;
}
out[0] += aa * yf;
out[1] += aa * uf;
out[2] += aa * vf;
}
}
}
if (s->h - H > 0) {
if (s->dmode == SEPARATE) {
for (n = 0; n < w; n++) {
float *cb = &s->combine_buffer[3 * n];
s->out->data[0][s->ypos * s->out->linesize[0] + n] = cb[0];
s->out->data[1][s->ypos * s->out->linesize[1] + n] = cb[1];
s->out->data[2][s->ypos * s->out->linesize[2] + n] = cb[2];
s->out->data[3][s->ypos * s->out->linesize[3] + n] = 255;
}
}
if (s->slide == SCROLL) {
for (p = 0; p < 4; p++) {
for (y = s->h; y >= H + 1; y--) {
memmove(s->out->data[p] + (y ) * s->out->linesize[p],
s->out->data[p] + (y-1) * s->out->linesize[p], w);
}
}
}
s->ypos++;
if (s->slide == SCROLL || s->ypos >= s->h)
s->ypos = H;
}
clone = av_frame_clone(s->out);
if (!clone)
return AVERROR(ENOMEM);
return ff_filter_frame(outlink, clone);
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AudioHistogramContext *s = ctx->priv;
AVFrame *in;
int ret;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in);
if (ret < 0)
return ret;
if (ret > 0)
return filter_frame(inlink, in);
FF_FILTER_FORWARD_STATUS(inlink, outlink);
FF_FILTER_FORWARD_WANTED(outlink, inlink);
return FFERROR_NOT_READY;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioHistogramContext *s = ctx->priv;
int i;
av_frame_free(&s->out);
av_freep(&s->shistogram);
av_freep(&s->achistogram);
av_freep(&s->combine_buffer);
for (i = 0; i < 101; i++)
av_frame_free(&s->in[i]);
}
static const AVFilterPad ahistogram_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
},
};
static const AVFilterPad ahistogram_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_VIDEO,
.config_props = config_output,
},
};
const AVFilter ff_avf_ahistogram = {
.name = "ahistogram",
.description = NULL_IF_CONFIG_SMALL("Convert input audio to histogram video output."),
.uninit = uninit,
.priv_size = sizeof(AudioHistogramContext),
.activate = activate,
FILTER_INPUTS(ahistogram_inputs),
FILTER_OUTPUTS(ahistogram_outputs),
FILTER_QUERY_FUNC(query_formats),
.priv_class = &ahistogram_class,
};