[FFmpeg-devel] [PATCH 2/2] lavfi: EBU R.128 scanner.
Stefano Sabatini
stefasab at gmail.com
Tue Sep 25 10:39:36 CEST 2012
On date Saturday 2012-09-22 10:16:02 +0200, Clément Bœsch encoded:
> TODO:
> - lavfi minor bump
> - Changelog
> ---
> configure | 1 +
> doc/filters.texi | 44 +++
> libavfilter/Makefile | 1 +
> libavfilter/af_ebur128.c | 689 +++++++++++++++++++++++++++++++++++++++++++++++
f_ebur128.c may be more suited, given the hybrid nature of the filter.
> libavfilter/allfilters.c | 1 +
> 5 files changed, 736 insertions(+)
> create mode 100644 libavfilter/af_ebur128.c
>
> diff --git a/configure b/configure
> index c001c5f..394ccf4 100755
> --- a/configure
> +++ b/configure
> @@ -1900,6 +1900,7 @@ decimate_filter_deps="gpl avcodec"
> delogo_filter_deps="gpl"
> deshake_filter_deps="avcodec"
> drawtext_filter_deps="libfreetype"
> +ebur128_filter_deps="gpl"
> flite_filter_deps="libflite"
> frei0r_filter_deps="frei0r dlopen"
> frei0r_filter_extralibs='$ldl'
> diff --git a/doc/filters.texi b/doc/filters.texi
> index e202d38..01f2778 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -4349,6 +4349,50 @@ setpts=PTS+10/TB
> @end example
> @end itemize
>
> + at section ebur128
> +
> +EBU R.128 scanner filter. This filter takes an audio stream as input and
> +outputs it unchanged. By default, it logs a message at a frequency of 10Hz with
> +the Momentary loudness (identified by @code{M}), Short-term loudness
> +(@code{S}), Integrated loudness (@code{I}) and Loudness Range (@code{LRA}).
Maybe add a link to the official spec document.
> +The filter also has a video output (see the @var{video} option) with a real
> +time graph to observe the loudness evolution. The graphic contains the logged
> +message mentioned above, so it is not printed anymore when this option is set,
> +unless the verbose logging is set. The main graphing area contains the
> +short-term loudness (3 seconds of analysis), and the gauge on the right is for
> +the momentary loudness (400 milliseconds).
> +
> +The filter accepts the following named parameters:
> +
> + at table @option
> +
> + at item video
> +Activate the video output. The audio stream is passed unchanged whether this
> +option is set or no. The video stream will appear in first position if
> +activated. Default is @code{0}.
Since the video is optional, I'd rather expect the video stream to
appear on the *second* output (and "appear in first position" is not
very clear).
> +
> + at item size
> +Set the video size. This option is for video only. Default and minimum
> +resolution is @code{640x480}.
> +
> + at item meter
> +Set the EBU scale meter. Default is @code{9}. Common values are @code{9} and
> + at code{18}, respectively for EBU scale meter +9 and EBU scale meter +18. Any
> +other integer value between this range is allowed.
> +
> + at end table
> +
> +Example of real-time graph using @command{ffplay}, with a EBU scale meter +18:
> + at example
> +ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]"
> + at end example
> +
> +Run an analysis with @command{ffmpeg}:
> + at example
> +ffmpeg -i input.mp3 -filter_complex ebur128 -f null -
> + at end example
> +
> @section settb, asettb
>
> Set the timebase to use for the output frames timestamps.
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 82d39e4..690d529 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -64,6 +64,7 @@ OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o
> OBJS-$(CONFIG_CHANNELMAP_FILTER) += af_channelmap.o
> OBJS-$(CONFIG_CHANNELSPLIT_FILTER) += af_channelsplit.o
> OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
> +OBJS-$(CONFIG_EBUR128_FILTER) += af_ebur128.o
> OBJS-$(CONFIG_JOIN_FILTER) += af_join.o
> OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
> OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
> diff --git a/libavfilter/af_ebur128.c b/libavfilter/af_ebur128.c
> new file mode 100644
> index 0000000..c9cfd46
> --- /dev/null
> +++ b/libavfilter/af_ebur128.c
> @@ -0,0 +1,689 @@
> +/*
> + * Copyright (c) 2012 Clément Bœsch
> + *
> + * This file is part of FFmpeg.
> + *
> + * FFmpeg is free software; you can redistribute it and/or modify
> + * it under the terms of the GNU General Public License as published by
> + * the Free Software Foundation; either version 2 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 General Public License for more details.
> + *
> + * You should have received a copy of the GNU 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
> + * EBU R.128 implementation
> + * @see http://tech.ebu.ch/loudness
> + * @see https://www.youtube.com/watch?v=iuEtQqC-Sqo "EBU R128 Introduction - Florian Camerer"
> + * @todo implement start/stop/reset through filter command injection
> + * @todo support other frequencies to avoid resampling
> + */
> +
> +#include <math.h>
> +
> +#include "libavutil/audioconvert.h"
> +#include "libavutil/avassert.h"
> +#include "libavutil/avstring.h"
> +#include "libavutil/xga_font_data.h"
> +#include "libavutil/opt.h"
> +#include "libavutil/timestamp.h"
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "formats.h"
> +#include "internal.h"
> +
> +#define MAX_CHANNELS 63
> +
> +/* pre-filter coefficients */
> +#define PRE_B0 1.53512485958697
> +#define PRE_B1 -2.69169618940638
> +#define PRE_B2 1.19839281085285
> +#define PRE_A1 -1.69065929318241
> +#define PRE_A2 0.73248077421585
> +
> +/* RLB-filter coefficients */
> +#define RLB_B0 1.0
> +#define RLB_B1 -2.0
> +#define RLB_B2 1.0
> +#define RLB_A1 -1.99004745483398
> +#define RLB_A2 0.99007225036621
> +
> +#define ABS_THRES -70 ///< silence gate: we discard anything below this absolute (LUFS) threshold
> +
> +struct integrator {
> + double *cache[MAX_CHANNELS]; ///< window of filtered samples (N ms)
> + int cache_pos; ///< focus on the last added bin in the cache array
> + double sum[MAX_CHANNELS]; ///< sum of the last N ms filtered samples (cache content)
> +
> + double sum_kept_powers; ///< sum of the powers (weighted sums) above absolute threshold
> + int nb_kept_powers; ///< number of sum above absolute threshold
> +
> +#define ABS_UP_THRES 10 ///< upper loud limit to consider (ABS_THRES being the minimum)
> +#define HIST_GRAIN 100 ///< defines histogram precision
> +#define HIST_SIZE ((ABS_UP_THRES - ABS_THRES) * HIST_GRAIN + 1)
> + int *powers_hist; ///< histogram of the powers, used to compute LRA and I
> +};
> +
> +struct rect { int x, y, w, h; };
> +
> +typedef struct {
> + const AVClass *class; ///< AVClass context for log and options purpose
> +
> + /* video */
> + int do_video; ///< 1 if video output enabled, 0 otherwise
> + int w, h; ///< size of the video output
> + struct rect text; ///< rectangle for the LU legend on the left
> + struct rect graph; ///< rectangle for the main graph in the center
> + struct rect gauge; ///< rectangle for the gauge on the right
> + AVFilterBufferRef *outpicref; ///< output picture reference, updated regularly
> + int meter; ///< select a EBU mode between +9 and +18
> + int scale_range; ///< the range of LU values according to the meter
> + int y_zero_lu; ///< the y value (pixel position) for 0 LU
> + int *y_line_ref; ///< y reference values for drawing the LU lines in the graph and the gauge
> +
> + /* audio */
> + int nb_channels; ///< number of channels in the input
> + double *ch_weighting; ///< channel weighting mapping
> + int sample_count; ///< sample count used for refresh frequency, reset at refresh
> +
> + /* Filter caches.
> + * The mult by 3 in the following is for X[i], X[i-1] and X[i-2] */
> + double x[MAX_CHANNELS * 3]; ///< 3 input samples cache for each channel
> + double y[MAX_CHANNELS * 3]; ///< 3 pre-filter samples cache for each channel
> + double z[MAX_CHANNELS * 3]; ///< 3 RLB-filter samples cache for each channel
> +
> +#define I400_BINS (48000 * 4 / 10)
> +#define I3000_BINS (48000 * 3)
> + struct integrator i400; ///< 400ms integrator, used for Momentary loudness (M), and Integrated loudness (I)
> + struct integrator i3000; ///< 3s integrator, used for Short term loudness (S), and Loudness Range (LRA)
> +} EBUR128Context;
> +
> +#define OFFSET(x) offsetof(EBUR128Context, x)
> +#define A AV_OPT_FLAG_AUDIO_PARAM
> +#define V AV_OPT_FLAG_VIDEO_PARAM
> +#define F AV_OPT_FLAG_FILTERING_PARAM
> +static const AVOption ebur128_options[] = {
> + { "video", "set video output", OFFSET(do_video), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, V|F },
> + { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "640x480"}, 0, 0, V|F },
> + { "meter", "set scale meter (+9 to +18)", OFFSET(meter), AV_OPT_TYPE_INT, {.i64 = 9}, 9, 18, V|F },
> + { NULL },
> +};
> +
> +AVFILTER_DEFINE_CLASS(ebur128);
> +
> +static const uint8_t graph_colors[] = {
> + 0xdd, 0x66, 0x66, // value above 0LU non reached
> + 0x66, 0x66, 0xdd, // value below 0LU non reached
> + 0x96, 0x33, 0x33, // value above 0LU reached
> + 0x33, 0x33, 0x96, // value below 0LU reached
> + 0xdd, 0x96, 0x96, // value above 0LU line non reached
> + 0x96, 0x96, 0xdd, // value below 0LU line non reached
> + 0xdd, 0x33, 0x33, // value above 0LU line reached
> + 0x33, 0x33, 0xdd, // value below 0LU line reached
> +};
Idle question: are those values specified by the spec?
> +static const uint8_t *get_graph_color(const EBUR128Context *ebur128, int v, int y)
> +{
> + const int below0 = y > ebur128->y_zero_lu;
> + const int reached = y >= v;
> + const int line = ebur128->y_line_ref[y] || y == ebur128->y_zero_lu;
> + const int colorid = 4*line + 2*reached + below0;
> + return graph_colors + 3*colorid;
> +}
> +
> +static inline int lu_to_y(const EBUR128Context *ebur128, double v)
> +{
> + v += 2 * ebur128->meter; // make it in range [0;...]
> + v = av_clipf(v, 0, ebur128->scale_range); // make sure it's in the graph scale
> + v = ebur128->scale_range - v; // invert value (y=0 is on top)
> + return v * ebur128->graph.h / ebur128->scale_range; // rescale from scale range to px height
> +}
> +
> +#define FONT8 0
> +#define FONT16 1
> +
> +static const uint8_t font_colors[] = {
> + 0xdd, 0xdd, 0x00,
> + 0x00, 0x96, 0x96,
> +};
> +
> +static void drawtext(AVFilterBufferRef *pic, int x, int y, int ftid, const uint8_t *color, const char *fmt, ...)
> +{
> + int i;
> + char buf[128] = {0};
> + const uint8_t *font;
> + int font_height;
> + va_list vl;
> +
> + if (ftid == FONT16) font = avpriv_vga16_font, font_height = 16;
> + else if (ftid == FONT8) font = avpriv_cga_font, font_height = 8;
> + else return;
> +
> + va_start(vl, fmt);
> + vsnprintf(buf, sizeof(buf), fmt, vl);
> + va_end(vl);
> +
> + for (i = 0; buf[i]; i++) {
> + int char_y, mask;
> + uint8_t *p = pic->data[0] + y*pic->linesize[0] + (x + i*8)*3;
> +
> + for (char_y = 0; char_y < font_height; char_y++) {
> + for (mask = 0x80; mask; mask >>= 1) {
> + if (font[buf[i] * font_height + char_y] & mask)
> + memcpy(p, color, 3);
> + else
> + memcpy(p, "\x00\x00\x00", 3);
> + p += 3;
> + }
> + p += pic->linesize[0] - 8*3;
> + }
> + }
> +}
> +
> +static void drawline(AVFilterBufferRef *pic, int x, int y, int len, int step)
> +{
> + int i;
> + uint8_t *p = pic->data[0] + y*pic->linesize[0] + x*3;
> +
> + for (i = 0; i < len; i++) {
> + memcpy(p, "\x00\xff\x00", 3);
> + p += step;
> + }
> +}
> +
> +static int config_video_output(AVFilterLink *outlink)
> +{
> + int i, x, y;
> + uint8_t *p;
> + AVFilterContext *ctx = outlink->src;
> + EBUR128Context *ebur128 = ctx->priv;
> + AVFilterBufferRef *outpicref;
> +
> + if (ebur128->w < 640 || ebur128->h < 480) {
> + av_log(ctx, AV_LOG_ERROR, "Video size %dx%d is too small, "
> + "minimum size is 640x480\n", ebur128->w, ebur128->h);
> + return AVERROR(EINVAL);
> + }
Are these constraints required by the standard? Or is there an
implementation-level reason for them?
> + outlink->w = ebur128->w;
> + outlink->h = ebur128->h;
> +
> +#define PAD 8
> +
> + /* configure text area position and size */
> + ebur128->text.x = PAD;
> + ebur128->text.y = 40;
> + ebur128->text.w = 3 * 8; // 3 characters
> + ebur128->text.h = ebur128->h - PAD - ebur128->text.y;
> +
> + /* configure gauge position and size */
> + ebur128->gauge.w = 20;
> + ebur128->gauge.h = ebur128->text.h;
> + ebur128->gauge.x = ebur128->w - PAD - ebur128->gauge.w;
> + ebur128->gauge.y = ebur128->text.y;
> +
> + /* configure graph position and size */
> + ebur128->graph.x = ebur128->text.x + ebur128->text.w + PAD;
> + ebur128->graph.y = ebur128->gauge.y;
> + ebur128->graph.w = ebur128->gauge.x - ebur128->graph.x - PAD;
> + ebur128->graph.h = ebur128->gauge.h;
> +
> + /* graph and gauge share the LU-to-pixel code */
> + av_assert0(ebur128->graph.h == ebur128->gauge.h);
> +
> + /* prepare the initial picref buffer */
> + avfilter_unref_bufferp(&ebur128->outpicref);
> + ebur128->outpicref = outpicref =
> + ff_get_video_buffer(outlink, AV_PERM_WRITE|AV_PERM_PRESERVE|AV_PERM_REUSE2,
> + outlink->w, outlink->h);
> + if (!outpicref)
> + return AVERROR(ENOMEM);
> + outlink->sample_aspect_ratio = (AVRational){1,1};
> +
> + /* init y references values (to draw LU lines) */
> + ebur128->y_line_ref = av_calloc(ebur128->graph.h + 1, sizeof(*ebur128->y_line_ref));
> + if (!ebur128->y_line_ref)
> + return AVERROR(ENOMEM);
> +
> + /* black background */
> + memset(outpicref->data[0], 0, ebur128->h * outpicref->linesize[0]);
> +
> + /* draw LU legends */
> + drawtext(outpicref, PAD, PAD+16, FONT8, font_colors+3, " LU");
> + for (i = ebur128->meter; i >= -ebur128->meter * 2; i--) {
> + y = lu_to_y(ebur128, i);
> + x = PAD + (i < 10 && i > -10) * 8;
> + ebur128->y_line_ref[y] = i;
> + y -= 4; // -4 to center vertically
> + drawtext(outpicref, x, y + ebur128->graph.y, FONT8, font_colors+3,
> + "%c%d", i < 0 ? '-' : i > 0 ? '+' : ' ', FFABS(i));
> + }
> +
> + /* draw graph */
> + ebur128->y_zero_lu = lu_to_y(ebur128, 0);
> + p = outpicref->data[0] + ebur128->graph.y * outpicref->linesize[0]
> + + ebur128->graph.x * 3;
> + for (y = 0; y < ebur128->graph.h; y++) {
> + const uint8_t *c = get_graph_color(ebur128, INT_MAX, y);
> +
> + for (x = 0; x < ebur128->graph.w; x++)
> + memcpy(p + x*3, c, 3);
> + p += outpicref->linesize[0];
> + }
> +
> + /* draw fancy rectangles around the graph and the gauge */
> +#define DRAW_RECT(r) do { \
> + drawline(outpicref, r.x, r.y - 1, r.w, 3); \
> + drawline(outpicref, r.x, r.y + r.h, r.w, 3); \
> + drawline(outpicref, r.x - 1, r.y, r.h, outpicref->linesize[0]); \
> + drawline(outpicref, r.x + r.w, r.y, r.h, outpicref->linesize[0]); \
> +} while (0)
> + DRAW_RECT(ebur128->graph);
> + DRAW_RECT(ebur128->gauge);
> +
> + return 0;
> +}
> +
> +static int config_audio_output(AVFilterLink *outlink)
> +{
> + int i;
> + AVFilterContext *ctx = outlink->src;
> + EBUR128Context *ebur128 = ctx->priv;
> + const int nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
> +
> +#define BACK_MASK (AV_CH_BACK_LEFT |AV_CH_BACK_CENTER |AV_CH_BACK_RIGHT| \
> + AV_CH_TOP_BACK_LEFT|AV_CH_TOP_BACK_CENTER|AV_CH_TOP_BACK_RIGHT)
> +
> + ebur128->nb_channels = nb_channels;
> + ebur128->ch_weighting = av_calloc(nb_channels, sizeof(*ebur128->ch_weighting));
> + if (!ebur128->ch_weighting)
> + return AVERROR(ENOMEM);
> +
> + for (i = 0; i < nb_channels; i++) {
> +
> + /* channel weighting */
> + if ((outlink->channel_layout & 1ULL<<i) == AV_CH_LOW_FREQUENCY)
> + continue;
> + if (outlink->channel_layout & 1ULL<<i & BACK_MASK)
> + ebur128->ch_weighting[i] = 1.41;
> + else
> + ebur128->ch_weighting[i] = 1.0;
> +
> + /* bins buffer for the two integration window (400ms and 3s) */
> + ebur128->i400.cache[i] = av_calloc(I400_BINS, sizeof(*ebur128->i400.cache[0]));
> + ebur128->i3000.cache[i] = av_calloc(I3000_BINS, sizeof(*ebur128->i3000.cache[0]));
> + if (!ebur128->i400.cache[i] || !ebur128->i3000.cache[i])
> + return AVERROR(ENOMEM);
> + }
> +
> + return 0;
> +}
> +
> +static av_cold int init(AVFilterContext *ctx, const char *args)
> +{
> + int ret;
> + EBUR128Context *ebur128 = ctx->priv;
> + AVFilterPad pad;
> +
> + ebur128->class = &ebur128_class;
> + av_opt_set_defaults(ebur128);
> +
> + if ((ret = av_set_options_string(ebur128, args, "=", ":")) < 0)
> + return ret;
> +
> + // if meter is +9 scale, scale range is from -18 LU to +9 LU (or 3*9)
> + // if meter is +18 scale, scale range is from -36 LU to +18 LU (or 3*18)
> + ebur128->scale_range = 3 * ebur128->meter;
> +
> + ebur128->i400.powers_hist = av_calloc(HIST_SIZE, sizeof(*ebur128->i400.powers_hist));
> + ebur128->i3000.powers_hist = av_calloc(HIST_SIZE, sizeof(*ebur128->i3000.powers_hist));
> +
> + /* insert output pads */
> + if (ebur128->do_video) {
> + pad = (AVFilterPad){
> + .name = av_strdup("out0"),
> + .type = AVMEDIA_TYPE_VIDEO,
> + .config_props = config_video_output,
> + };
> + if (!pad.name)
> + return AVERROR(ENOMEM);
> + ff_insert_outpad(ctx, 0, &pad);
> + }
> + pad = (AVFilterPad){
> + .name = av_asprintf("out%d", ebur128->do_video),
> + .type = AVMEDIA_TYPE_AUDIO,
> + .config_props = config_audio_output,
> + };
> + if (!pad.name)
> + return AVERROR(ENOMEM);
> + ff_insert_outpad(ctx, ebur128->do_video, &pad);
> +
> + /* summary */
> + av_log(ctx, AV_LOG_VERBOSE, "EBU +%d scale\n", ebur128->meter);
> +
> + return 0;
> +}
> +
> +#define HIST_POS(power) (int)(((power) - ABS_THRES) * HIST_GRAIN)
> +#define HIST_POW(pos) ((double)(pos) / HIST_GRAIN + ABS_THRES)
> +
> +/* loudness and power should be set such as loudness = -0.691 +
> + * 10*log10(power), we just avoid doing that calculus two times */
> +static int gate_update(struct integrator *integ, double power,
> + double loudness, int gate_thres)
> +{
> + int ipower;
> + double relative_threshold;
> + int gate_hist_pos;
> +
> + /* update powers histograms by incrementing current power count */
> + ipower = av_clip(HIST_POS(loudness), 0, HIST_SIZE - 1);
> + integ->powers_hist[ipower]++;
> +
> + /* compute relative threshold and get its position in the histogram */
> + integ->sum_kept_powers += power;
> + integ->nb_kept_powers++;
> + relative_threshold = integ->sum_kept_powers / integ->nb_kept_powers;
> + relative_threshold = -0.691 + 10*log10(relative_threshold) + gate_thres;
> + gate_hist_pos = av_clip(HIST_POS(relative_threshold), 0, HIST_SIZE - 1);
> +
> + return gate_hist_pos;
> +}
> +
> +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
> +{
> + int i, ch;
> + AVFilterContext *ctx = inlink->dst;
> + EBUR128Context *ebur128 = ctx->priv;
> + const int nb_channels = ebur128->nb_channels;
> + const int nb_samples = insamples->audio->nb_samples;
> + const double *samples = (double *)insamples->data[0];
> + AVFilterBufferRef *pic = ebur128->outpicref;
> +
> + for (i = 0; i < nb_samples; i++) {
> +
> + const int bin_id_400 = ebur128->i400.cache_pos;
> + const int bin_id_3000 = ebur128->i3000.cache_pos;
> +
> + if (++ebur128->i400.cache_pos == I400_BINS) ebur128->i400.cache_pos = 0;
> + if (++ebur128->i3000.cache_pos == I3000_BINS) ebur128->i3000.cache_pos = 0;
> +
> + for (ch = 0; ch < nb_channels; ch++) {
> + double bin;
> +
> + if (!ebur128->ch_weighting[ch])
> + continue;
> +
> + /* Y[i] = X[i]*b0 + X[i-1]*b1 + X[i-2]*b2 - Y[i-1]*a1 - Y[i-2]*a2 */
> +#define FILTER(Y, X, name) do { \
> + double *dst = ebur128->Y + ch*3; \
> + double *src = ebur128->X + ch*3; \
> + dst[2] = dst[1]; \
> + dst[1] = dst[0]; \
> + dst[0] = src[0]*name##_B0 + src[1]*name##_B1 + src[2]*name##_B2 \
> + - dst[1]*name##_A1 - dst[2]*name##_A2; \
> +} while (0)
> +
> + ebur128->x[ch * 3] = *samples++; // set X[i]
> +
> + // TODO: merge both filters in one?
> + FILTER(y, x, PRE); // apply pre-filter
> + ebur128->x[ch * 3 + 2] = ebur128->x[ch * 3 + 1];
> + ebur128->x[ch * 3 + 1] = ebur128->x[ch * 3 ];
> + FILTER(z, y, RLB); // apply RLB-filter
> +
> + bin = ebur128->z[ch * 3] * ebur128->z[ch * 3];
> +
> + /* add the new value, and limit the sum to the cache size (400ms or 3s)
> + * by removing the oldest one */
> + ebur128->i400.sum [ch] = ebur128->i400.sum [ch] + bin - ebur128->i400.cache [ch][bin_id_400];
> + ebur128->i3000.sum[ch] = ebur128->i3000.sum[ch] + bin - ebur128->i3000.cache[ch][bin_id_3000];
> +
> + /* override old cache entry with the new value */
> + ebur128->i400.cache [ch][bin_id_400 ] = bin;
> + ebur128->i3000.cache[ch][bin_id_3000] = bin;
> + }
> +
> + /* For integrated loudness, gating blocks are 400ms long with 75%
> + * overlap (see BS.1770-2 p5), so a recomputation is needed each 100ms
> + * (4800 samples at 48kHz). */
> + if (++ebur128->sample_count == 4800) {
> + double loudness_400, loudness_3000;
> + double power_400 = 0, power_3000 = 0;
> + double integrated_loudness = 0;
> + double loudness_range = 0;
> +
> + ebur128->sample_count = 0;
> +
> +#define COMPUTE_LOUDNESS(m, time) do { \
> + /* weighting sum of the last <time> ms */ \
> + for (ch = 0; ch < nb_channels; ch++) \
> + power_##time += ebur128->ch_weighting[ch] * ebur128->i##time.sum[ch]; \
> + power_##time /= I##time##_BINS; \
> + \
> + loudness_##time = -0.691 + 10*log10(power_##time); \
> +} while (0)
> +
> + COMPUTE_LOUDNESS(M, 400);
> + COMPUTE_LOUDNESS(S, 3000);
> +
> + /* Integrated loudness */
> +#define I_GATE_THRES -10 // initially defined to -8 LU in the first EBU standard
> +
> + if (loudness_400 >= ABS_THRES) {
> + double integrated_sum = 0;
> + int nb_integrated = 0;
> + int gate_hist_pos = gate_update(&ebur128->i400, power_400,
> + loudness_400, I_GATE_THRES);
> +
> + /* compute integrated loudness by summing the histogram values
> + * above the relative threshold */
> + for (i = gate_hist_pos; i < HIST_SIZE; i++) {
> + const int nb_v = ebur128->i400.powers_hist[i];
> + if (nb_v) {
> + nb_integrated += nb_v;
> + integrated_sum += nb_v * HIST_POW(i); // XXX: cache pow in histogram?
> + }
> + }
> + if (nb_integrated)
> + integrated_loudness = integrated_sum / nb_integrated;
> + }
> +
> + /* LRA */
> +#define LRA_GATE_THRES -20
> +#define LRA_LOWER_PRC 10
> +#define LRA_HIGHER_PRC 95
> +
> + /* XXX: example code in EBU 3342 is ">=" but formula in BS.1770
> + * specs is ">" */
> + if (loudness_3000 >= ABS_THRES) {
> + int nb_powers = 0;
> + int gate_hist_pos = gate_update(&ebur128->i3000, power_3000,
> + loudness_3000, LRA_GATE_THRES);
> +
> + for (i = gate_hist_pos; i < HIST_SIZE; i++)
> + nb_powers += ebur128->i3000.powers_hist[i];
> + if (nb_powers) {
> + int n, nb_pow;
> + double low_v = 0, high_v = 0;
> +
> + /* get lower power to consider */
> + n = 0;
> + nb_pow = LRA_LOWER_PRC * nb_powers / 100. + 0.5;
> + for (i = gate_hist_pos; i < HIST_SIZE; i++) {
> + n += ebur128->i3000.powers_hist[i];
> + if (n >= nb_pow) {
> + low_v = HIST_POW(i);
> + break;
> + }
> + }
> +
> + /* get higher power to consider */
> + n = nb_powers;
> + nb_pow = LRA_HIGHER_PRC * nb_powers / 100. + 0.5;
> + for (i = HIST_SIZE - 1; i >= 0; i--) {
> + n -= ebur128->i3000.powers_hist[i];
> + if (n < nb_pow) {
> + high_v = HIST_POW(i);
> + break;
> + }
> + }
> +
> + // TODO: show low & high on the graph
> + loudness_range = high_v - low_v;
> + }
> + }
> +
> +#define LOG_FMT "M:%6.1f S:%6.1f I:%6.1f LUFS LRA:%6.1f LU"
> +
> + /* push one video frame */
> + if (ebur128->do_video) {
> + int x, y;
> + uint8_t *p;
> + AVFilterLink *outlink = ctx->outputs[0];
> +
> + const int y_loudness_lu_graph = lu_to_y(ebur128, loudness_3000 + 23);
> + const int y_loudness_lu_gauge = lu_to_y(ebur128, loudness_400 + 23);
> +
> + /* draw the graph using the short-term loudness */
> + p = pic->data[0] + ebur128->graph.y*pic->linesize[0] + ebur128->graph.x*3;
> + for (y = 0; y < ebur128->graph.h; y++) {
> + const uint8_t *c = get_graph_color(ebur128, y_loudness_lu_graph, y);
> +
> + memmove(p, p + 3, (ebur128->graph.w - 1) * 3);
> + memcpy(p + (ebur128->graph.w - 1) * 3, c, 3);
> + p += pic->linesize[0];
> + }
> +
> + /* draw the gauge using the momentary loudness */
> + p = pic->data[0] + ebur128->gauge.y*pic->linesize[0] + ebur128->gauge.x*3;
> + for (y = 0; y < ebur128->gauge.h; y++) {
> + const uint8_t *c = get_graph_color(ebur128, y_loudness_lu_gauge, y);
> +
> + for (x = 0; x < ebur128->gauge.w; x++)
> + memcpy(p + x*3, c, 3);
> + p += pic->linesize[0];
> + }
> +
> + /* draw textual info */
> + drawtext(pic, PAD, PAD - PAD/2, FONT16, font_colors,
> + LOG_FMT " ", // padding to erase trailing characters
> + loudness_400, loudness_3000,
> + integrated_loudness, loudness_range);
> +
> + /* set pts and push frame */
> + pic->pts = insamples->pts +
> + av_rescale_q(i, (AVRational){ 1, inlink->sample_rate },
> + outlink->time_base);
> + ff_start_frame(outlink, avfilter_ref_buffer(pic, ~AV_PERM_WRITE));
> + ff_draw_slice(outlink, 0, outlink->h, 1);
> + ff_end_frame(outlink);
missing checks on return values
> + }
> +
> + av_log(ctx, ebur128->do_video ? AV_LOG_VERBOSE : AV_LOG_INFO,
> + LOG_FMT "\n", loudness_400, loudness_3000,
> + integrated_loudness, loudness_range);
> + }
> + }
> +
> + return ff_filter_samples(ctx->outputs[ebur128->do_video], insamples);
> +}
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> + EBUR128Context *ebur128 = ctx->priv;
> + AVFilterFormats *formats;
> + AVFilterChannelLayouts *layouts;
> + AVFilterLink *inlink = ctx->inputs[0];
> + AVFilterLink *outlink = ctx->outputs[0];
> +
> + static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_DBL, -1 };
> + static const int input_srate[] = {48000, -1}; // ITU-R BS.1770 provides coeff only for 48kHz
> + static const enum PixelFormat pix_fmts[] = { PIX_FMT_RGB24, -1 };
> +
> + /* set input audio formats */
> + formats = ff_make_format_list(sample_fmts);
> + if (!formats)
> + return AVERROR(ENOMEM);
> + ff_formats_ref(formats, &inlink->out_formats);
> +
> + layouts = ff_all_channel_layouts();
> + if (!layouts)
> + return AVERROR(ENOMEM);
> + ff_channel_layouts_ref(layouts, &inlink->out_channel_layouts);
> +
> + formats = ff_make_format_list(input_srate);
> + if (!formats)
> + return AVERROR(ENOMEM);
> + ff_formats_ref(formats, &inlink->out_samplerates);
> +
> + /* set optional output video format */
> + if (ebur128->do_video) {
> + formats = ff_make_format_list(pix_fmts);
> + if (!formats)
> + return AVERROR(ENOMEM);
> + ff_formats_ref(formats, &outlink->in_formats);
> + outlink = ctx->outputs[1];
> + }
> +
> + /* set audio output formats (same as input since it's just a passthrough) */
> + formats = ff_make_format_list(sample_fmts);
> + if (!formats)
> + return AVERROR(ENOMEM);
> + ff_formats_ref(formats, &outlink->in_formats);
> +
> + layouts = ff_all_channel_layouts();
> + if (!layouts)
> + return AVERROR(ENOMEM);
> + ff_channel_layouts_ref(layouts, &outlink->in_channel_layouts);
> +
> + formats = ff_make_format_list(input_srate);
> + if (!formats)
> + return AVERROR(ENOMEM);
> + ff_formats_ref(formats, &outlink->in_samplerates);
> +
> + return 0;
> +}
This could be defined more logically before filter_samples().
> +static av_cold void uninit(AVFilterContext *ctx)
> +{
> + int i;
> + EBUR128Context *ebur128 = ctx->priv;
> +
> + av_freep(&ebur128->y_line_ref);
> + av_freep(&ebur128->ch_weighting);
> + av_freep(&ebur128->i400.powers_hist);
> + av_freep(&ebur128->i3000.powers_hist);
> + for (i = 0; i < ebur128->nb_channels; i++) {
> + av_freep(&ebur128->i400.cache[i]);
> + av_freep(&ebur128->i3000.cache[i]);
> + }
> + for (i = 0; i < ctx->nb_outputs; i++)
> + av_freep(&ctx->output_pads[i].name);
> + avfilter_unref_bufferp(&ebur128->outpicref);
> +}
> +
> +AVFilter avfilter_af_ebur128 = {
> + .name = "ebur128",
> + .description = NULL_IF_CONFIG_SMALL("EBU R.128 scanner."),
> + .priv_size = sizeof(EBUR128Context),
> + .init = init,
> + .uninit = uninit,
> + .query_formats = query_formats,
> +
> + .inputs = (const AVFilterPad[]) {
> + { .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .get_audio_buffer = ff_null_get_audio_buffer,
> + .filter_samples = filter_samples, },
> + { .name = NULL }
> + },
> + .outputs = (const AVFilterPad[]) { { .name = NULL } },
> +};
Looks great otherwise, I didn't read it very carefully, but I think we
can commit and let it mature then.
I also wonder if we could make it a bit more flexible, for example to
make the short-term loudness and the momentary loudness windows
configurable (possibly useful for other purposes), but I don't
consider this blocking.
--
FFmpeg = Fundamental & Fanciful Mournful Powerful Enhanced Genius
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