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FFmpeg/libavfilter/avf_aphasemeter.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

397 lines
14 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
*/
/**
* @file
* audio to video multimedia aphasemeter filter
*/
#include "libavutil/channel_layout.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/opt.h"
#include "libavutil/parseutils.h"
#include "libavutil/timestamp.h"
#include "avfilter.h"
#include "formats.h"
#include "audio.h"
#include "video.h"
#include "internal.h"
#include "float.h"
typedef struct AudioPhaseMeterContext {
const AVClass *class;
AVFrame *out;
int do_video;
int do_phasing_detection;
int w, h;
AVRational frame_rate;
int contrast[4];
uint8_t *mpc_str;
uint8_t mpc[4];
int draw_median_phase;
int is_mono;
int is_out_phase;
int start_mono_presence;
int start_out_phase_presence;
float tolerance;
float angle;
float phase;
AVRational time_base;
int64_t duration;
int64_t frame_end;
int64_t mono_idx[2];
int64_t out_phase_idx[2];
} AudioPhaseMeterContext;
#define MAX_DURATION (24*60*60*1000000LL)
#define OFFSET(x) offsetof(AudioPhaseMeterContext, x)
#define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_VIDEO_PARAM
#define get_duration(index) (index[1] - index[0])
static const AVOption aphasemeter_options[] = {
{ "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="800x400"}, 0, 0, FLAGS },
{ "s", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str="800x400"}, 0, 0, FLAGS },
{ "rc", "set red contrast", OFFSET(contrast[0]), AV_OPT_TYPE_INT, {.i64=2}, 0, 255, FLAGS },
{ "gc", "set green contrast", OFFSET(contrast[1]), AV_OPT_TYPE_INT, {.i64=7}, 0, 255, FLAGS },
{ "bc", "set blue contrast", OFFSET(contrast[2]), AV_OPT_TYPE_INT, {.i64=1}, 0, 255, FLAGS },
{ "mpc", "set median phase color", OFFSET(mpc_str), AV_OPT_TYPE_STRING, {.str = "none"}, 0, 0, FLAGS },
{ "video", "set video output", OFFSET(do_video), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
{ "phasing", "set mono and out-of-phase detection output", OFFSET(do_phasing_detection), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
{ "tolerance", "set phase tolerance for mono detection", OFFSET(tolerance), AV_OPT_TYPE_FLOAT, {.dbl = 0.}, 0, 1, FLAGS },
{ "t", "set phase tolerance for mono detection", OFFSET(tolerance), AV_OPT_TYPE_FLOAT, {.dbl = 0.}, 0, 1, FLAGS },
{ "angle", "set angle threshold for out-of-phase detection", OFFSET(angle), AV_OPT_TYPE_FLOAT, {.dbl = 170.}, 90, 180, FLAGS },
{ "a", "set angle threshold for out-of-phase detection", OFFSET(angle), AV_OPT_TYPE_FLOAT, {.dbl = 170.}, 90, 180, FLAGS },
{ "duration", "set minimum mono or out-of-phase duration in seconds", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64=2000000}, 0, MAX_DURATION, FLAGS },
{ "d", "set minimum mono or out-of-phase duration in seconds", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64=2000000}, 0, MAX_DURATION, FLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(aphasemeter);
static int query_formats(AVFilterContext *ctx)
{
AudioPhaseMeterContext *s = ctx->priv;
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layout = NULL;
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_NONE };
static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_RGBA, AV_PIX_FMT_NONE };
int ret;
formats = ff_make_format_list(sample_fmts);
if ((ret = ff_formats_ref (formats, &inlink->outcfg.formats )) < 0 ||
(ret = ff_formats_ref (formats, &outlink->incfg.formats )) < 0 ||
(ret = ff_add_channel_layout (&layout, AV_CH_LAYOUT_STEREO )) < 0 ||
(ret = ff_channel_layouts_ref (layout , &inlink->outcfg.channel_layouts)) < 0 ||
(ret = ff_channel_layouts_ref (layout , &outlink->incfg.channel_layouts)) < 0)
return ret;
formats = ff_all_samplerates();
if ((ret = ff_formats_ref(formats, &inlink->outcfg.samplerates)) < 0 ||
(ret = ff_formats_ref(formats, &outlink->incfg.samplerates)) < 0)
return ret;
if (s->do_video) {
AVFilterLink *outlink = ctx->outputs[1];
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;
AudioPhaseMeterContext *s = ctx->priv;
int nb_samples;
s->duration = av_rescale(s->duration, inlink->sample_rate, AV_TIME_BASE);
if (s->do_video) {
nb_samples = FFMAX(1, av_rescale(inlink->sample_rate, s->frame_rate.den, s->frame_rate.num));
inlink->min_samples =
inlink->max_samples = nb_samples;
}
return 0;
}
static int config_video_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioPhaseMeterContext *s = ctx->priv;
outlink->w = s->w;
outlink->h = s->h;
outlink->sample_aspect_ratio = (AVRational){1,1};
outlink->frame_rate = s->frame_rate;
if (!strcmp(s->mpc_str, "none"))
s->draw_median_phase = 0;
else if (av_parse_color(s->mpc, s->mpc_str, -1, ctx) >= 0)
s->draw_median_phase = 1;
else
return AVERROR(EINVAL);
return 0;
}
static inline int get_x(float phase, int w)
{
return (phase + 1.) / 2. * (w - 1);
}
static inline void add_metadata(AVFrame *insamples, const char *key, char *value)
{
char buf[128];
snprintf(buf, sizeof(buf), "lavfi.aphasemeter.%s", key);
av_dict_set(&insamples->metadata, buf, value, 0);
}
static inline void update_mono_detection(AudioPhaseMeterContext *s, AVFrame *insamples, int mono_measurement)
{
int64_t mono_duration;
if (!s->is_mono && mono_measurement) {
s->is_mono = 1;
s->start_mono_presence = 1;
s->mono_idx[0] = insamples->pts;
}
if (s->is_mono && mono_measurement && s->start_mono_presence) {
s->mono_idx[1] = s->frame_end;
mono_duration = get_duration(s->mono_idx);
if (mono_duration >= s->duration) {
add_metadata(insamples, "mono_start", av_ts2timestr(s->mono_idx[0], &s->time_base));
av_log(s, AV_LOG_INFO, "mono_start: %s\n", av_ts2timestr(s->mono_idx[0], &s->time_base));
s->start_mono_presence = 0;
}
}
if (s->is_mono && !mono_measurement) {
s->mono_idx[1] = insamples ? insamples->pts : s->frame_end;
mono_duration = get_duration(s->mono_idx);
if (mono_duration >= s->duration) {
if (insamples) {
add_metadata(insamples, "mono_end", av_ts2timestr(s->mono_idx[1], &s->time_base));
add_metadata(insamples, "mono_duration", av_ts2timestr(mono_duration, &s->time_base));
}
av_log(s, AV_LOG_INFO, "mono_end: %s | mono_duration: %s\n", av_ts2timestr(s->mono_idx[1], &s->time_base), av_ts2timestr(mono_duration, &s->time_base));
}
s->is_mono = 0;
}
}
static inline void update_out_phase_detection(AudioPhaseMeterContext *s, AVFrame *insamples, int out_phase_measurement)
{
int64_t out_phase_duration;
if (!s->is_out_phase && out_phase_measurement) {
s->is_out_phase = 1;
s->start_out_phase_presence = 1;
s->out_phase_idx[0] = insamples->pts;
}
if (s->is_out_phase && out_phase_measurement && s->start_out_phase_presence) {
s->out_phase_idx[1] = s->frame_end;
out_phase_duration = get_duration(s->out_phase_idx);
if (out_phase_duration >= s->duration) {
add_metadata(insamples, "out_phase_start", av_ts2timestr(s->out_phase_idx[0], &s->time_base));
av_log(s, AV_LOG_INFO, "out_phase_start: %s\n", av_ts2timestr(s->out_phase_idx[0], &s->time_base));
s->start_out_phase_presence = 0;
}
}
if (s->is_out_phase && !out_phase_measurement) {
s->out_phase_idx[1] = insamples ? insamples->pts : s->frame_end;
out_phase_duration = get_duration(s->out_phase_idx);
if (out_phase_duration >= s->duration) {
if (insamples) {
add_metadata(insamples, "out_phase_end", av_ts2timestr(s->out_phase_idx[1], &s->time_base));
add_metadata(insamples, "out_phase_duration", av_ts2timestr(out_phase_duration, &s->time_base));
}
av_log(s, AV_LOG_INFO, "out_phase_end: %s | out_phase_duration: %s\n", av_ts2timestr(s->out_phase_idx[1], &s->time_base), av_ts2timestr(out_phase_duration, &s->time_base));
}
s->is_out_phase = 0;
}
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AudioPhaseMeterContext *s = ctx->priv;
AVFilterLink *outlink = s->do_video ? ctx->outputs[1] : NULL;
AVFilterLink *aoutlink = ctx->outputs[0];
AVDictionary **metadata;
const int rc = s->contrast[0];
const int gc = s->contrast[1];
const int bc = s->contrast[2];
float fphase = 0;
AVFrame *out;
uint8_t *dst;
int i;
int mono_measurement;
int out_phase_measurement;
float tolerance = 1.0f - s->tolerance;
float angle = cosf(s->angle/180.0f*M_PI);
if (s->do_video && (!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);
}
out = s->out;
for (i = 0; i < outlink->h; i++)
memset(out->data[0] + i * out->linesize[0], 0, outlink->w * 4);
} else if (s->do_video) {
out = s->out;
for (i = outlink->h - 1; i >= 10; i--)
memmove(out->data[0] + (i ) * out->linesize[0],
out->data[0] + (i-1) * out->linesize[0],
outlink->w * 4);
for (i = 0; i < outlink->w; i++)
AV_WL32(out->data[0] + i * 4, 0);
}
for (i = 0; i < in->nb_samples; i++) {
const float *src = (float *)in->data[0] + i * 2;
const float f = src[0] * src[1] / (src[0]*src[0] + src[1] * src[1]) * 2;
const float phase = isnan(f) ? 1 : f;
const int x = get_x(phase, s->w);
if (s->do_video) {
dst = out->data[0] + x * 4;
dst[0] = FFMIN(255, dst[0] + rc);
dst[1] = FFMIN(255, dst[1] + gc);
dst[2] = FFMIN(255, dst[2] + bc);
dst[3] = 255;
}
fphase += phase;
}
fphase /= in->nb_samples;
s->phase = fphase;
if (s->do_video) {
if (s->draw_median_phase) {
dst = out->data[0] + get_x(fphase, s->w) * 4;
AV_WL32(dst, AV_RL32(s->mpc));
}
for (i = 1; i < 10 && i < outlink->h; i++)
memcpy(out->data[0] + i * out->linesize[0], out->data[0], outlink->w * 4);
}
metadata = &in->metadata;
if (metadata) {
uint8_t value[128];
snprintf(value, sizeof(value), "%f", fphase);
add_metadata(in, "phase", value);
}
if (s->do_phasing_detection) {
s->time_base = inlink->time_base;
s->frame_end = in->pts + av_rescale_q(in->nb_samples,
(AVRational){ 1, in->sample_rate }, inlink->time_base);
mono_measurement = (tolerance - fphase) < FLT_EPSILON;
out_phase_measurement = (angle - fphase) > FLT_EPSILON;
update_mono_detection(s, in, mono_measurement);
update_out_phase_detection(s, in, out_phase_measurement);
}
if (s->do_video) {
AVFrame *clone;
s->out->pts = in->pts;
clone = av_frame_clone(s->out);
if (!clone)
return AVERROR(ENOMEM);
ff_filter_frame(outlink, clone);
}
return ff_filter_frame(aoutlink, in);
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioPhaseMeterContext *s = ctx->priv;
if (s->do_phasing_detection) {
update_mono_detection(s, NULL, 0);
update_out_phase_detection(s, NULL, 0);
}
av_frame_free(&s->out);
}
static av_cold int init(AVFilterContext *ctx)
{
AudioPhaseMeterContext *s = ctx->priv;
AVFilterPad pad;
int ret;
pad = (AVFilterPad){
.name = "out0",
.type = AVMEDIA_TYPE_AUDIO,
};
ret = ff_append_outpad(ctx, &pad);
if (ret < 0)
return ret;
if (s->do_video) {
pad = (AVFilterPad){
.name = "out1",
.type = AVMEDIA_TYPE_VIDEO,
.config_props = config_video_output,
};
ret = ff_append_outpad(ctx, &pad);
if (ret < 0)
return ret;
}
return 0;
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
.filter_frame = filter_frame,
},
};
const AVFilter ff_avf_aphasemeter = {
.name = "aphasemeter",
.description = NULL_IF_CONFIG_SMALL("Convert input audio to phase meter video output."),
.init = init,
.uninit = uninit,
.priv_size = sizeof(AudioPhaseMeterContext),
FILTER_INPUTS(inputs),
.outputs = NULL,
FILTER_QUERY_FUNC(query_formats),
.priv_class = &aphasemeter_class,
.flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS,
};