From: Paul B Mahol <onemda@gmail.com>
To: FFmpeg development discussions and patches <ffmpeg-devel@ffmpeg.org>
Subject: [FFmpeg-devel] [PATCH] avfilter: merge loudnorm filter functionality into f_ebur128.c
Date: Fri, 29 Sep 2023 23:46:07 +0200
Message-ID: <CAPYw7P7AE0pZSzCOKXOF5DjABa8fJbOGLtReBBW5mB1D5=X9aQ@mail.gmail.com> (raw)
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From 4f0023bda8b34a005aa32a1ff03ebc32a2d65bda Mon Sep 17 00:00:00 2001
From: Paul B Mahol <onemda@gmail.com>
Date: Fri, 29 Sep 2023 20:53:51 +0200
Subject: [PATCH] avfilter: merge loudnorm filter functionality into
f_ebur128.c
Removes ebur128 - an old version of library and replaces it with native implementation.
Not complete, going to add currently missing dynamic filtering mode.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
libavfilter/Makefile | 1 -
libavfilter/af_loudnorm.c | 941 --------------------------------------
libavfilter/ebur128.c | 725 -----------------------------
libavfilter/ebur128.h | 229 ----------
libavfilter/f_ebur128.c | 720 ++++++++++++++++++++++-------
5 files changed, 559 insertions(+), 2057 deletions(-)
delete mode 100644 libavfilter/af_loudnorm.c
delete mode 100644 libavfilter/ebur128.c
delete mode 100644 libavfilter/ebur128.h
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 45ef50a30c..cefebea59f 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -150,7 +150,6 @@ OBJS-$(CONFIG_HIGHPASS_FILTER) += af_biquads.o
OBJS-$(CONFIG_HIGHSHELF_FILTER) += af_biquads.o
OBJS-$(CONFIG_JOIN_FILTER) += af_join.o
OBJS-$(CONFIG_LADSPA_FILTER) += af_ladspa.o
-OBJS-$(CONFIG_LOUDNORM_FILTER) += af_loudnorm.o ebur128.o
OBJS-$(CONFIG_LOWPASS_FILTER) += af_biquads.o
OBJS-$(CONFIG_LOWSHELF_FILTER) += af_biquads.o
OBJS-$(CONFIG_LV2_FILTER) += af_lv2.o
diff --git a/libavfilter/af_loudnorm.c b/libavfilter/af_loudnorm.c
deleted file mode 100644
index d83398ae2a..0000000000
--- a/libavfilter/af_loudnorm.c
+++ /dev/null
@@ -1,941 +0,0 @@
-/*
- * Copyright (c) 2016 Kyle Swanson <k@ylo.ph>.
- *
- * 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
- */
-
-/* http://k.ylo.ph/2016/04/04/loudnorm.html */
-
-#include "libavutil/opt.h"
-#include "avfilter.h"
-#include "filters.h"
-#include "formats.h"
-#include "internal.h"
-#include "audio.h"
-#include "ebur128.h"
-
-enum FrameType {
- FIRST_FRAME,
- INNER_FRAME,
- FINAL_FRAME,
- LINEAR_MODE,
- FRAME_NB
-};
-
-enum LimiterState {
- OUT,
- ATTACK,
- SUSTAIN,
- RELEASE,
- STATE_NB
-};
-
-enum PrintFormat {
- NONE,
- JSON,
- SUMMARY,
- PF_NB
-};
-
-typedef struct LoudNormContext {
- const AVClass *class;
- double target_i;
- double target_lra;
- double target_tp;
- double measured_i;
- double measured_lra;
- double measured_tp;
- double measured_thresh;
- double offset;
- int linear;
- int dual_mono;
- enum PrintFormat print_format;
-
- double *buf;
- int buf_size;
- int buf_index;
- int prev_buf_index;
-
- double delta[30];
- double weights[21];
- double prev_delta;
- int index;
-
- double gain_reduction[2];
- double *limiter_buf;
- double *prev_smp;
- int limiter_buf_index;
- int limiter_buf_size;
- enum LimiterState limiter_state;
- int peak_index;
- int env_index;
- int env_cnt;
- int attack_length;
- int release_length;
-
- int64_t pts[30];
- enum FrameType frame_type;
- int above_threshold;
- int prev_nb_samples;
- int channels;
-
- FFEBUR128State *r128_in;
- FFEBUR128State *r128_out;
-} LoudNormContext;
-
-#define OFFSET(x) offsetof(LoudNormContext, x)
-#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
-
-static const AVOption loudnorm_options[] = {
- { "I", "set integrated loudness target", OFFSET(target_i), AV_OPT_TYPE_DOUBLE, {.dbl = -24.}, -70., -5., FLAGS },
- { "i", "set integrated loudness target", OFFSET(target_i), AV_OPT_TYPE_DOUBLE, {.dbl = -24.}, -70., -5., FLAGS },
- { "LRA", "set loudness range target", OFFSET(target_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 7.}, 1., 50., FLAGS },
- { "lra", "set loudness range target", OFFSET(target_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 7.}, 1., 50., FLAGS },
- { "TP", "set maximum true peak", OFFSET(target_tp), AV_OPT_TYPE_DOUBLE, {.dbl = -2.}, -9., 0., FLAGS },
- { "tp", "set maximum true peak", OFFSET(target_tp), AV_OPT_TYPE_DOUBLE, {.dbl = -2.}, -9., 0., FLAGS },
- { "measured_I", "measured IL of input file", OFFSET(measured_i), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 0., FLAGS },
- { "measured_i", "measured IL of input file", OFFSET(measured_i), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 0., FLAGS },
- { "measured_LRA", "measured LRA of input file", OFFSET(measured_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, 0., 99., FLAGS },
- { "measured_lra", "measured LRA of input file", OFFSET(measured_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, 0., 99., FLAGS },
- { "measured_TP", "measured true peak of input file", OFFSET(measured_tp), AV_OPT_TYPE_DOUBLE, {.dbl = 99.}, -99., 99., FLAGS },
- { "measured_tp", "measured true peak of input file", OFFSET(measured_tp), AV_OPT_TYPE_DOUBLE, {.dbl = 99.}, -99., 99., FLAGS },
- { "measured_thresh", "measured threshold of input file", OFFSET(measured_thresh), AV_OPT_TYPE_DOUBLE, {.dbl = -70.}, -99., 0., FLAGS },
- { "offset", "set offset gain", OFFSET(offset), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 99., FLAGS },
- { "linear", "normalize linearly if possible", OFFSET(linear), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
- { "dual_mono", "treat mono input as dual-mono", OFFSET(dual_mono), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
- { "print_format", "set print format for stats", OFFSET(print_format), AV_OPT_TYPE_INT, {.i64 = NONE}, NONE, PF_NB -1, FLAGS, "print_format" },
- { "none", 0, 0, AV_OPT_TYPE_CONST, {.i64 = NONE}, 0, 0, FLAGS, "print_format" },
- { "json", 0, 0, AV_OPT_TYPE_CONST, {.i64 = JSON}, 0, 0, FLAGS, "print_format" },
- { "summary", 0, 0, AV_OPT_TYPE_CONST, {.i64 = SUMMARY}, 0, 0, FLAGS, "print_format" },
- { NULL }
-};
-
-AVFILTER_DEFINE_CLASS(loudnorm);
-
-static inline int frame_size(int sample_rate, int frame_len_msec)
-{
- const int frame_size = round((double)sample_rate * (frame_len_msec / 1000.0));
- return frame_size + (frame_size % 2);
-}
-
-static void init_gaussian_filter(LoudNormContext *s)
-{
- double total_weight = 0.0;
- const double sigma = 3.5;
- double adjust;
- int i;
-
- const int offset = 21 / 2;
- const double c1 = 1.0 / (sigma * sqrt(2.0 * M_PI));
- const double c2 = 2.0 * pow(sigma, 2.0);
-
- for (i = 0; i < 21; i++) {
- const int x = i - offset;
- s->weights[i] = c1 * exp(-(pow(x, 2.0) / c2));
- total_weight += s->weights[i];
- }
-
- adjust = 1.0 / total_weight;
- for (i = 0; i < 21; i++)
- s->weights[i] *= adjust;
-}
-
-static double gaussian_filter(LoudNormContext *s, int index)
-{
- double result = 0.;
- int i;
-
- index = index - 10 > 0 ? index - 10 : index + 20;
- for (i = 0; i < 21; i++)
- result += s->delta[((index + i) < 30) ? (index + i) : (index + i - 30)] * s->weights[i];
-
- return result;
-}
-
-static void detect_peak(LoudNormContext *s, int offset, int nb_samples, int channels, int *peak_delta, double *peak_value)
-{
- int n, c, i, index;
- double ceiling;
- double *buf;
-
- *peak_delta = -1;
- buf = s->limiter_buf;
- ceiling = s->target_tp;
-
- index = s->limiter_buf_index + (offset * channels) + (1920 * channels);
- if (index >= s->limiter_buf_size)
- index -= s->limiter_buf_size;
-
- if (s->frame_type == FIRST_FRAME) {
- for (c = 0; c < channels; c++)
- s->prev_smp[c] = fabs(buf[index + c - channels]);
- }
-
- for (n = 0; n < nb_samples; n++) {
- for (c = 0; c < channels; c++) {
- double this, next, max_peak;
-
- this = fabs(buf[(index + c) < s->limiter_buf_size ? (index + c) : (index + c - s->limiter_buf_size)]);
- next = fabs(buf[(index + c + channels) < s->limiter_buf_size ? (index + c + channels) : (index + c + channels - s->limiter_buf_size)]);
-
- if ((s->prev_smp[c] <= this) && (next <= this) && (this > ceiling) && (n > 0)) {
- int detected;
-
- detected = 1;
- for (i = 2; i < 12; i++) {
- next = fabs(buf[(index + c + (i * channels)) < s->limiter_buf_size ? (index + c + (i * channels)) : (index + c + (i * channels) - s->limiter_buf_size)]);
- if (next > this) {
- detected = 0;
- break;
- }
- }
-
- if (!detected)
- continue;
-
- for (c = 0; c < channels; c++) {
- if (c == 0 || fabs(buf[index + c]) > max_peak)
- max_peak = fabs(buf[index + c]);
-
- s->prev_smp[c] = fabs(buf[(index + c) < s->limiter_buf_size ? (index + c) : (index + c - s->limiter_buf_size)]);
- }
-
- *peak_delta = n;
- s->peak_index = index;
- *peak_value = max_peak;
- return;
- }
-
- s->prev_smp[c] = this;
- }
-
- index += channels;
- if (index >= s->limiter_buf_size)
- index -= s->limiter_buf_size;
- }
-}
-
-static void true_peak_limiter(LoudNormContext *s, double *out, int nb_samples, int channels)
-{
- int n, c, index, peak_delta, smp_cnt;
- double ceiling, peak_value;
- double *buf;
-
- buf = s->limiter_buf;
- ceiling = s->target_tp;
- index = s->limiter_buf_index;
- smp_cnt = 0;
-
- if (s->frame_type == FIRST_FRAME) {
- double max;
-
- max = 0.;
- for (n = 0; n < 1920; n++) {
- for (c = 0; c < channels; c++) {
- max = fabs(buf[c]) > max ? fabs(buf[c]) : max;
- }
- buf += channels;
- }
-
- if (max > ceiling) {
- s->gain_reduction[1] = ceiling / max;
- s->limiter_state = SUSTAIN;
- buf = s->limiter_buf;
-
- for (n = 0; n < 1920; n++) {
- for (c = 0; c < channels; c++) {
- double env;
- env = s->gain_reduction[1];
- buf[c] *= env;
- }
- buf += channels;
- }
- }
-
- buf = s->limiter_buf;
- }
-
- do {
-
- switch(s->limiter_state) {
- case OUT:
- detect_peak(s, smp_cnt, nb_samples - smp_cnt, channels, &peak_delta, &peak_value);
- if (peak_delta != -1) {
- s->env_cnt = 0;
- smp_cnt += (peak_delta - s->attack_length);
- s->gain_reduction[0] = 1.;
- s->gain_reduction[1] = ceiling / peak_value;
- s->limiter_state = ATTACK;
-
- s->env_index = s->peak_index - (s->attack_length * channels);
- if (s->env_index < 0)
- s->env_index += s->limiter_buf_size;
-
- s->env_index += (s->env_cnt * channels);
- if (s->env_index > s->limiter_buf_size)
- s->env_index -= s->limiter_buf_size;
-
- } else {
- smp_cnt = nb_samples;
- }
- break;
-
- case ATTACK:
- for (; s->env_cnt < s->attack_length; s->env_cnt++) {
- for (c = 0; c < channels; c++) {
- double env;
- env = s->gain_reduction[0] - ((double) s->env_cnt / (s->attack_length - 1) * (s->gain_reduction[0] - s->gain_reduction[1]));
- buf[s->env_index + c] *= env;
- }
-
- s->env_index += channels;
- if (s->env_index >= s->limiter_buf_size)
- s->env_index -= s->limiter_buf_size;
-
- smp_cnt++;
- if (smp_cnt >= nb_samples) {
- s->env_cnt++;
- break;
- }
- }
-
- if (smp_cnt < nb_samples) {
- s->env_cnt = 0;
- s->attack_length = 1920;
- s->limiter_state = SUSTAIN;
- }
- break;
-
- case SUSTAIN:
- detect_peak(s, smp_cnt, nb_samples, channels, &peak_delta, &peak_value);
- if (peak_delta == -1) {
- s->limiter_state = RELEASE;
- s->gain_reduction[0] = s->gain_reduction[1];
- s->gain_reduction[1] = 1.;
- s->env_cnt = 0;
- break;
- } else {
- double gain_reduction;
- gain_reduction = ceiling / peak_value;
-
- if (gain_reduction < s->gain_reduction[1]) {
- s->limiter_state = ATTACK;
-
- s->attack_length = peak_delta;
- if (s->attack_length <= 1)
- s->attack_length = 2;
-
- s->gain_reduction[0] = s->gain_reduction[1];
- s->gain_reduction[1] = gain_reduction;
- s->env_cnt = 0;
- break;
- }
-
- for (s->env_cnt = 0; s->env_cnt < peak_delta; s->env_cnt++) {
- for (c = 0; c < channels; c++) {
- double env;
- env = s->gain_reduction[1];
- buf[s->env_index + c] *= env;
- }
-
- s->env_index += channels;
- if (s->env_index >= s->limiter_buf_size)
- s->env_index -= s->limiter_buf_size;
-
- smp_cnt++;
- if (smp_cnt >= nb_samples) {
- s->env_cnt++;
- break;
- }
- }
- }
- break;
-
- case RELEASE:
- for (; s->env_cnt < s->release_length; s->env_cnt++) {
- for (c = 0; c < channels; c++) {
- double env;
- env = s->gain_reduction[0] + (((double) s->env_cnt / (s->release_length - 1)) * (s->gain_reduction[1] - s->gain_reduction[0]));
- buf[s->env_index + c] *= env;
- }
-
- s->env_index += channels;
- if (s->env_index >= s->limiter_buf_size)
- s->env_index -= s->limiter_buf_size;
-
- smp_cnt++;
- if (smp_cnt >= nb_samples) {
- s->env_cnt++;
- break;
- }
- }
-
- if (smp_cnt < nb_samples) {
- s->env_cnt = 0;
- s->limiter_state = OUT;
- }
-
- break;
- }
-
- } while (smp_cnt < nb_samples);
-
- for (n = 0; n < nb_samples; n++) {
- for (c = 0; c < channels; c++) {
- out[c] = buf[index + c];
- if (fabs(out[c]) > ceiling) {
- out[c] = ceiling * (out[c] < 0 ? -1 : 1);
- }
- }
- out += channels;
- index += channels;
- if (index >= s->limiter_buf_size)
- index -= s->limiter_buf_size;
- }
-}
-
-static int filter_frame(AVFilterLink *inlink, AVFrame *in)
-{
- AVFilterContext *ctx = inlink->dst;
- LoudNormContext *s = ctx->priv;
- AVFilterLink *outlink = ctx->outputs[0];
- AVFrame *out;
- const double *src;
- double *dst;
- double *buf;
- double *limiter_buf;
- int i, n, c, subframe_length, src_index;
- double gain, gain_next, env_global, env_shortterm,
- global, shortterm, lra, relative_threshold;
-
- if (av_frame_is_writable(in)) {
- out = in;
- } else {
- out = ff_get_audio_buffer(outlink, in->nb_samples);
- if (!out) {
- av_frame_free(&in);
- return AVERROR(ENOMEM);
- }
- av_frame_copy_props(out, in);
- }
-
- out->pts = s->pts[0];
- memmove(s->pts, &s->pts[1], (FF_ARRAY_ELEMS(s->pts) - 1) * sizeof(s->pts[0]));
-
- src = (const double *)in->data[0];
- dst = (double *)out->data[0];
- buf = s->buf;
- limiter_buf = s->limiter_buf;
-
- ff_ebur128_add_frames_double(s->r128_in, src, in->nb_samples);
-
- if (s->frame_type == FIRST_FRAME && in->nb_samples < frame_size(inlink->sample_rate, 3000)) {
- double offset, offset_tp, true_peak;
-
- ff_ebur128_loudness_global(s->r128_in, &global);
- for (c = 0; c < inlink->ch_layout.nb_channels; c++) {
- double tmp;
- ff_ebur128_sample_peak(s->r128_in, c, &tmp);
- if (c == 0 || tmp > true_peak)
- true_peak = tmp;
- }
-
- offset = pow(10., (s->target_i - global) / 20.);
- offset_tp = true_peak * offset;
- s->offset = offset_tp < s->target_tp ? offset : s->target_tp / true_peak;
- s->frame_type = LINEAR_MODE;
- }
-
- switch (s->frame_type) {
- case FIRST_FRAME:
- for (n = 0; n < in->nb_samples; n++) {
- for (c = 0; c < inlink->ch_layout.nb_channels; c++) {
- buf[s->buf_index + c] = src[c];
- }
- src += inlink->ch_layout.nb_channels;
- s->buf_index += inlink->ch_layout.nb_channels;
- }
-
- ff_ebur128_loudness_shortterm(s->r128_in, &shortterm);
-
- if (shortterm < s->measured_thresh) {
- s->above_threshold = 0;
- env_shortterm = shortterm <= -70. ? 0. : s->target_i - s->measured_i;
- } else {
- s->above_threshold = 1;
- env_shortterm = shortterm <= -70. ? 0. : s->target_i - shortterm;
- }
-
- for (n = 0; n < 30; n++)
- s->delta[n] = pow(10., env_shortterm / 20.);
- s->prev_delta = s->delta[s->index];
-
- s->buf_index =
- s->limiter_buf_index = 0;
-
- for (n = 0; n < (s->limiter_buf_size / inlink->ch_layout.nb_channels); n++) {
- for (c = 0; c < inlink->ch_layout.nb_channels; c++) {
- limiter_buf[s->limiter_buf_index + c] = buf[s->buf_index + c] * s->delta[s->index] * s->offset;
- }
- s->limiter_buf_index += inlink->ch_layout.nb_channels;
- if (s->limiter_buf_index >= s->limiter_buf_size)
- s->limiter_buf_index -= s->limiter_buf_size;
-
- s->buf_index += inlink->ch_layout.nb_channels;
- }
-
- subframe_length = frame_size(inlink->sample_rate, 100);
- true_peak_limiter(s, dst, subframe_length, inlink->ch_layout.nb_channels);
- ff_ebur128_add_frames_double(s->r128_out, dst, subframe_length);
-
- out->nb_samples = subframe_length;
-
- s->frame_type = INNER_FRAME;
- break;
-
- case INNER_FRAME:
- gain = gaussian_filter(s, s->index + 10 < 30 ? s->index + 10 : s->index + 10 - 30);
- gain_next = gaussian_filter(s, s->index + 11 < 30 ? s->index + 11 : s->index + 11 - 30);
-
- for (n = 0; n < in->nb_samples; n++) {
- for (c = 0; c < inlink->ch_layout.nb_channels; c++) {
- buf[s->prev_buf_index + c] = src[c];
- limiter_buf[s->limiter_buf_index + c] = buf[s->buf_index + c] * (gain + (((double) n / in->nb_samples) * (gain_next - gain))) * s->offset;
- }
- src += inlink->ch_layout.nb_channels;
-
- s->limiter_buf_index += inlink->ch_layout.nb_channels;
- if (s->limiter_buf_index >= s->limiter_buf_size)
- s->limiter_buf_index -= s->limiter_buf_size;
-
- s->prev_buf_index += inlink->ch_layout.nb_channels;
- if (s->prev_buf_index >= s->buf_size)
- s->prev_buf_index -= s->buf_size;
-
- s->buf_index += inlink->ch_layout.nb_channels;
- if (s->buf_index >= s->buf_size)
- s->buf_index -= s->buf_size;
- }
-
- subframe_length = (frame_size(inlink->sample_rate, 100) - in->nb_samples) * inlink->ch_layout.nb_channels;
- s->limiter_buf_index = s->limiter_buf_index + subframe_length < s->limiter_buf_size ? s->limiter_buf_index + subframe_length : s->limiter_buf_index + subframe_length - s->limiter_buf_size;
-
- true_peak_limiter(s, dst, in->nb_samples, inlink->ch_layout.nb_channels);
- ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples);
-
- ff_ebur128_loudness_range(s->r128_in, &lra);
- ff_ebur128_loudness_global(s->r128_in, &global);
- ff_ebur128_loudness_shortterm(s->r128_in, &shortterm);
- ff_ebur128_relative_threshold(s->r128_in, &relative_threshold);
-
- if (s->above_threshold == 0) {
- double shortterm_out;
-
- if (shortterm > s->measured_thresh)
- s->prev_delta *= 1.0058;
-
- ff_ebur128_loudness_shortterm(s->r128_out, &shortterm_out);
- if (shortterm_out >= s->target_i)
- s->above_threshold = 1;
- }
-
- if (shortterm < relative_threshold || shortterm <= -70. || s->above_threshold == 0) {
- s->delta[s->index] = s->prev_delta;
- } else {
- env_global = fabs(shortterm - global) < (s->target_lra / 2.) ? shortterm - global : (s->target_lra / 2.) * ((shortterm - global) < 0 ? -1 : 1);
- env_shortterm = s->target_i - shortterm;
- s->delta[s->index] = pow(10., (env_global + env_shortterm) / 20.);
- }
-
- s->prev_delta = s->delta[s->index];
- s->index++;
- if (s->index >= 30)
- s->index -= 30;
- s->prev_nb_samples = in->nb_samples;
- break;
-
- case FINAL_FRAME:
- gain = gaussian_filter(s, s->index + 10 < 30 ? s->index + 10 : s->index + 10 - 30);
- s->limiter_buf_index = 0;
- src_index = 0;
-
- for (n = 0; n < s->limiter_buf_size / inlink->ch_layout.nb_channels; n++) {
- for (c = 0; c < inlink->ch_layout.nb_channels; c++) {
- s->limiter_buf[s->limiter_buf_index + c] = src[src_index + c] * gain * s->offset;
- }
- src_index += inlink->ch_layout.nb_channels;
-
- s->limiter_buf_index += inlink->ch_layout.nb_channels;
- if (s->limiter_buf_index >= s->limiter_buf_size)
- s->limiter_buf_index -= s->limiter_buf_size;
- }
-
- subframe_length = frame_size(inlink->sample_rate, 100);
- for (i = 0; i < in->nb_samples / subframe_length; i++) {
- true_peak_limiter(s, dst, subframe_length, inlink->ch_layout.nb_channels);
-
- for (n = 0; n < subframe_length; n++) {
- for (c = 0; c < inlink->ch_layout.nb_channels; c++) {
- if (src_index < (in->nb_samples * inlink->ch_layout.nb_channels)) {
- limiter_buf[s->limiter_buf_index + c] = src[src_index + c] * gain * s->offset;
- } else {
- limiter_buf[s->limiter_buf_index + c] = 0.;
- }
- }
-
- if (src_index < (in->nb_samples * inlink->ch_layout.nb_channels))
- src_index += inlink->ch_layout.nb_channels;
-
- s->limiter_buf_index += inlink->ch_layout.nb_channels;
- if (s->limiter_buf_index >= s->limiter_buf_size)
- s->limiter_buf_index -= s->limiter_buf_size;
- }
-
- dst += (subframe_length * inlink->ch_layout.nb_channels);
- }
-
- dst = (double *)out->data[0];
- ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples);
- break;
-
- case LINEAR_MODE:
- for (n = 0; n < in->nb_samples; n++) {
- for (c = 0; c < inlink->ch_layout.nb_channels; c++) {
- dst[c] = src[c] * s->offset;
- }
- src += inlink->ch_layout.nb_channels;
- dst += inlink->ch_layout.nb_channels;
- }
-
- dst = (double *)out->data[0];
- ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples);
- break;
- }
-
- if (in != out)
- av_frame_free(&in);
- return ff_filter_frame(outlink, out);
-}
-
-static int flush_frame(AVFilterLink *outlink)
-{
- AVFilterContext *ctx = outlink->src;
- AVFilterLink *inlink = ctx->inputs[0];
- LoudNormContext *s = ctx->priv;
- int ret = 0;
-
- if (s->frame_type == INNER_FRAME) {
- double *src;
- double *buf;
- int nb_samples, n, c, offset;
- AVFrame *frame;
-
- nb_samples = (s->buf_size / inlink->ch_layout.nb_channels) - s->prev_nb_samples;
- nb_samples -= (frame_size(inlink->sample_rate, 100) - s->prev_nb_samples);
-
- frame = ff_get_audio_buffer(outlink, nb_samples);
- if (!frame)
- return AVERROR(ENOMEM);
- frame->nb_samples = nb_samples;
-
- buf = s->buf;
- src = (double *)frame->data[0];
-
- offset = ((s->limiter_buf_size / inlink->ch_layout.nb_channels) - s->prev_nb_samples) * inlink->ch_layout.nb_channels;
- offset -= (frame_size(inlink->sample_rate, 100) - s->prev_nb_samples) * inlink->ch_layout.nb_channels;
- s->buf_index = s->buf_index - offset < 0 ? s->buf_index - offset + s->buf_size : s->buf_index - offset;
-
- for (n = 0; n < nb_samples; n++) {
- for (c = 0; c < inlink->ch_layout.nb_channels; c++) {
- src[c] = buf[s->buf_index + c];
- }
- src += inlink->ch_layout.nb_channels;
- s->buf_index += inlink->ch_layout.nb_channels;
- if (s->buf_index >= s->buf_size)
- s->buf_index -= s->buf_size;
- }
-
- s->frame_type = FINAL_FRAME;
- ret = filter_frame(inlink, frame);
- }
- return ret;
-}
-
-static int activate(AVFilterContext *ctx)
-{
- AVFilterLink *inlink = ctx->inputs[0];
- AVFilterLink *outlink = ctx->outputs[0];
- LoudNormContext *s = ctx->priv;
- AVFrame *in = NULL;
- int ret = 0, status;
- int64_t pts;
-
- FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
-
- if (s->frame_type != LINEAR_MODE) {
- int nb_samples;
-
- if (s->frame_type == FIRST_FRAME) {
- nb_samples = frame_size(inlink->sample_rate, 3000);
- } else {
- nb_samples = frame_size(inlink->sample_rate, 100);
- }
-
- ret = ff_inlink_consume_samples(inlink, nb_samples, nb_samples, &in);
- } else {
- ret = ff_inlink_consume_frame(inlink, &in);
- }
-
- if (ret < 0)
- return ret;
- if (ret > 0) {
- if (s->frame_type == FIRST_FRAME) {
- const int nb_samples = frame_size(inlink->sample_rate, 100);
-
- for (int i = 0; i < FF_ARRAY_ELEMS(s->pts); i++)
- s->pts[i] = in->pts + i * nb_samples;
- } else if (s->frame_type == LINEAR_MODE) {
- s->pts[0] = in->pts;
- } else {
- s->pts[FF_ARRAY_ELEMS(s->pts) - 1] = in->pts;
- }
- ret = filter_frame(inlink, in);
- }
- if (ret < 0)
- return ret;
-
- if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
- ff_outlink_set_status(outlink, status, pts);
- return flush_frame(outlink);
- }
-
- FF_FILTER_FORWARD_WANTED(outlink, inlink);
-
- return FFERROR_NOT_READY;
-}
-
-static int query_formats(AVFilterContext *ctx)
-{
- LoudNormContext *s = ctx->priv;
- static const int input_srate[] = {192000, -1};
- static const enum AVSampleFormat sample_fmts[] = {
- AV_SAMPLE_FMT_DBL,
- AV_SAMPLE_FMT_NONE
- };
- int ret = ff_set_common_all_channel_counts(ctx);
- if (ret < 0)
- return ret;
-
- ret = ff_set_common_formats_from_list(ctx, sample_fmts);
- if (ret < 0)
- return ret;
-
- if (s->frame_type == LINEAR_MODE) {
- return ff_set_common_all_samplerates(ctx);
- } else {
- return ff_set_common_samplerates_from_list(ctx, input_srate);
- }
-}
-
-static int config_input(AVFilterLink *inlink)
-{
- AVFilterContext *ctx = inlink->dst;
- LoudNormContext *s = ctx->priv;
-
- s->r128_in = ff_ebur128_init(inlink->ch_layout.nb_channels, inlink->sample_rate, 0, FF_EBUR128_MODE_I | FF_EBUR128_MODE_S | FF_EBUR128_MODE_LRA | FF_EBUR128_MODE_SAMPLE_PEAK);
- if (!s->r128_in)
- return AVERROR(ENOMEM);
-
- s->r128_out = ff_ebur128_init(inlink->ch_layout.nb_channels, inlink->sample_rate, 0, FF_EBUR128_MODE_I | FF_EBUR128_MODE_S | FF_EBUR128_MODE_LRA | FF_EBUR128_MODE_SAMPLE_PEAK);
- if (!s->r128_out)
- return AVERROR(ENOMEM);
-
- if (inlink->ch_layout.nb_channels == 1 && s->dual_mono) {
- ff_ebur128_set_channel(s->r128_in, 0, FF_EBUR128_DUAL_MONO);
- ff_ebur128_set_channel(s->r128_out, 0, FF_EBUR128_DUAL_MONO);
- }
-
- s->buf_size = frame_size(inlink->sample_rate, 3000) * inlink->ch_layout.nb_channels;
- s->buf = av_malloc_array(s->buf_size, sizeof(*s->buf));
- if (!s->buf)
- return AVERROR(ENOMEM);
-
- s->limiter_buf_size = frame_size(inlink->sample_rate, 210) * inlink->ch_layout.nb_channels;
- s->limiter_buf = av_malloc_array(s->buf_size, sizeof(*s->limiter_buf));
- if (!s->limiter_buf)
- return AVERROR(ENOMEM);
-
- s->prev_smp = av_malloc_array(inlink->ch_layout.nb_channels, sizeof(*s->prev_smp));
- if (!s->prev_smp)
- return AVERROR(ENOMEM);
-
- init_gaussian_filter(s);
-
- s->buf_index =
- s->prev_buf_index =
- s->limiter_buf_index = 0;
- s->channels = inlink->ch_layout.nb_channels;
- s->index = 1;
- s->limiter_state = OUT;
- s->offset = pow(10., s->offset / 20.);
- s->target_tp = pow(10., s->target_tp / 20.);
- s->attack_length = frame_size(inlink->sample_rate, 10);
- s->release_length = frame_size(inlink->sample_rate, 100);
-
- return 0;
-}
-
-static av_cold int init(AVFilterContext *ctx)
-{
- LoudNormContext *s = ctx->priv;
- s->frame_type = FIRST_FRAME;
-
- if (s->linear) {
- double offset, offset_tp;
- offset = s->target_i - s->measured_i;
- offset_tp = s->measured_tp + offset;
-
- if (s->measured_tp != 99 && s->measured_thresh != -70 && s->measured_lra != 0 && s->measured_i != 0) {
- if ((offset_tp <= s->target_tp) && (s->measured_lra <= s->target_lra)) {
- s->frame_type = LINEAR_MODE;
- s->offset = offset;
- }
- }
- }
-
- return 0;
-}
-
-static av_cold void uninit(AVFilterContext *ctx)
-{
- LoudNormContext *s = ctx->priv;
- double i_in, i_out, lra_in, lra_out, thresh_in, thresh_out, tp_in, tp_out;
- int c;
-
- if (!s->r128_in || !s->r128_out)
- goto end;
-
- ff_ebur128_loudness_range(s->r128_in, &lra_in);
- ff_ebur128_loudness_global(s->r128_in, &i_in);
- ff_ebur128_relative_threshold(s->r128_in, &thresh_in);
- for (c = 0; c < s->channels; c++) {
- double tmp;
- ff_ebur128_sample_peak(s->r128_in, c, &tmp);
- if ((c == 0) || (tmp > tp_in))
- tp_in = tmp;
- }
-
- ff_ebur128_loudness_range(s->r128_out, &lra_out);
- ff_ebur128_loudness_global(s->r128_out, &i_out);
- ff_ebur128_relative_threshold(s->r128_out, &thresh_out);
- for (c = 0; c < s->channels; c++) {
- double tmp;
- ff_ebur128_sample_peak(s->r128_out, c, &tmp);
- if ((c == 0) || (tmp > tp_out))
- tp_out = tmp;
- }
-
- switch(s->print_format) {
- case NONE:
- break;
-
- case JSON:
- av_log(ctx, AV_LOG_INFO,
- "\n{\n"
- "\t\"input_i\" : \"%.2f\",\n"
- "\t\"input_tp\" : \"%.2f\",\n"
- "\t\"input_lra\" : \"%.2f\",\n"
- "\t\"input_thresh\" : \"%.2f\",\n"
- "\t\"output_i\" : \"%.2f\",\n"
- "\t\"output_tp\" : \"%+.2f\",\n"
- "\t\"output_lra\" : \"%.2f\",\n"
- "\t\"output_thresh\" : \"%.2f\",\n"
- "\t\"normalization_type\" : \"%s\",\n"
- "\t\"target_offset\" : \"%.2f\"\n"
- "}\n",
- i_in,
- 20. * log10(tp_in),
- lra_in,
- thresh_in,
- i_out,
- 20. * log10(tp_out),
- lra_out,
- thresh_out,
- s->frame_type == LINEAR_MODE ? "linear" : "dynamic",
- s->target_i - i_out
- );
- break;
-
- case SUMMARY:
- av_log(ctx, AV_LOG_INFO,
- "\n"
- "Input Integrated: %+6.1f LUFS\n"
- "Input True Peak: %+6.1f dBTP\n"
- "Input LRA: %6.1f LU\n"
- "Input Threshold: %+6.1f LUFS\n"
- "\n"
- "Output Integrated: %+6.1f LUFS\n"
- "Output True Peak: %+6.1f dBTP\n"
- "Output LRA: %6.1f LU\n"
- "Output Threshold: %+6.1f LUFS\n"
- "\n"
- "Normalization Type: %s\n"
- "Target Offset: %+6.1f LU\n",
- i_in,
- 20. * log10(tp_in),
- lra_in,
- thresh_in,
- i_out,
- 20. * log10(tp_out),
- lra_out,
- thresh_out,
- s->frame_type == LINEAR_MODE ? "Linear" : "Dynamic",
- s->target_i - i_out
- );
- break;
- }
-
-end:
- if (s->r128_in)
- ff_ebur128_destroy(&s->r128_in);
- if (s->r128_out)
- ff_ebur128_destroy(&s->r128_out);
- av_freep(&s->limiter_buf);
- av_freep(&s->prev_smp);
- av_freep(&s->buf);
-}
-
-static const AVFilterPad avfilter_af_loudnorm_inputs[] = {
- {
- .name = "default",
- .type = AVMEDIA_TYPE_AUDIO,
- .config_props = config_input,
- },
-};
-
-const AVFilter ff_af_loudnorm = {
- .name = "loudnorm",
- .description = NULL_IF_CONFIG_SMALL("EBU R128 loudness normalization"),
- .priv_size = sizeof(LoudNormContext),
- .priv_class = &loudnorm_class,
- .init = init,
- .activate = activate,
- .uninit = uninit,
- FILTER_INPUTS(avfilter_af_loudnorm_inputs),
- FILTER_OUTPUTS(ff_audio_default_filterpad),
- FILTER_QUERY_FUNC(query_formats),
-};
diff --git a/libavfilter/ebur128.c b/libavfilter/ebur128.c
deleted file mode 100644
index 062099e206..0000000000
--- a/libavfilter/ebur128.c
+++ /dev/null
@@ -1,725 +0,0 @@
-/*
- * Copyright (c) 2011 Jan Kokemüller
- *
- * 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
- *
- * This file is based on libebur128 which is available at
- * https://github.com/jiixyj/libebur128/
- *
- * Libebur128 has the following copyright:
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
-*/
-
-#include "ebur128.h"
-
-#include <float.h>
-#include <limits.h>
-#include <math.h> /* You may have to define _USE_MATH_DEFINES if you use MSVC */
-
-#include "libavutil/error.h"
-#include "libavutil/macros.h"
-#include "libavutil/mem.h"
-#include "libavutil/mem_internal.h"
-#include "libavutil/thread.h"
-
-#define CHECK_ERROR(condition, errorcode, goto_point) \
- if ((condition)) { \
- errcode = (errorcode); \
- goto goto_point; \
- }
-
-#define ALMOST_ZERO 0.000001
-
-#define RELATIVE_GATE (-10.0)
-#define RELATIVE_GATE_FACTOR pow(10.0, RELATIVE_GATE / 10.0)
-#define MINUS_20DB pow(10.0, -20.0 / 10.0)
-
-struct FFEBUR128StateInternal {
- /** Filtered audio data (used as ring buffer). */
- double *audio_data;
- /** Size of audio_data array. */
- size_t audio_data_frames;
- /** Current index for audio_data. */
- size_t audio_data_index;
- /** How many frames are needed for a gating block. Will correspond to 400ms
- * of audio at initialization, and 100ms after the first block (75% overlap
- * as specified in the 2011 revision of BS1770). */
- unsigned long needed_frames;
- /** The channel map. Has as many elements as there are channels. */
- int *channel_map;
- /** How many samples fit in 100ms (rounded). */
- unsigned long samples_in_100ms;
- /** BS.1770 filter coefficients (nominator). */
- double b[5];
- /** BS.1770 filter coefficients (denominator). */
- double a[5];
- /** BS.1770 filter state. */
- double v[5][5];
- /** Histograms, used to calculate LRA. */
- unsigned long *block_energy_histogram;
- unsigned long *short_term_block_energy_histogram;
- /** Keeps track of when a new short term block is needed. */
- size_t short_term_frame_counter;
- /** Maximum sample peak, one per channel */
- double *sample_peak;
- /** The maximum window duration in ms. */
- unsigned long window;
- /** Data pointer array for interleaved data */
- void **data_ptrs;
-};
-
-static AVOnce histogram_init = AV_ONCE_INIT;
-static DECLARE_ALIGNED(32, double, histogram_energies)[1000];
-static DECLARE_ALIGNED(32, double, histogram_energy_boundaries)[1001];
-
-static void ebur128_init_filter(FFEBUR128State * st)
-{
- int i, j;
-
- double f0 = 1681.974450955533;
- double G = 3.999843853973347;
- double Q = 0.7071752369554196;
-
- double K = tan(M_PI * f0 / (double) st->samplerate);
- double Vh = pow(10.0, G / 20.0);
- double Vb = pow(Vh, 0.4996667741545416);
-
- double pb[3] = { 0.0, 0.0, 0.0 };
- double pa[3] = { 1.0, 0.0, 0.0 };
- double rb[3] = { 1.0, -2.0, 1.0 };
- double ra[3] = { 1.0, 0.0, 0.0 };
-
- double a0 = 1.0 + K / Q + K * K;
- pb[0] = (Vh + Vb * K / Q + K * K) / a0;
- pb[1] = 2.0 * (K * K - Vh) / a0;
- pb[2] = (Vh - Vb * K / Q + K * K) / a0;
- pa[1] = 2.0 * (K * K - 1.0) / a0;
- pa[2] = (1.0 - K / Q + K * K) / a0;
-
- f0 = 38.13547087602444;
- Q = 0.5003270373238773;
- K = tan(M_PI * f0 / (double) st->samplerate);
-
- ra[1] = 2.0 * (K * K - 1.0) / (1.0 + K / Q + K * K);
- ra[2] = (1.0 - K / Q + K * K) / (1.0 + K / Q + K * K);
-
- st->d->b[0] = pb[0] * rb[0];
- st->d->b[1] = pb[0] * rb[1] + pb[1] * rb[0];
- st->d->b[2] = pb[0] * rb[2] + pb[1] * rb[1] + pb[2] * rb[0];
- st->d->b[3] = pb[1] * rb[2] + pb[2] * rb[1];
- st->d->b[4] = pb[2] * rb[2];
-
- st->d->a[0] = pa[0] * ra[0];
- st->d->a[1] = pa[0] * ra[1] + pa[1] * ra[0];
- st->d->a[2] = pa[0] * ra[2] + pa[1] * ra[1] + pa[2] * ra[0];
- st->d->a[3] = pa[1] * ra[2] + pa[2] * ra[1];
- st->d->a[4] = pa[2] * ra[2];
-
- for (i = 0; i < 5; ++i) {
- for (j = 0; j < 5; ++j) {
- st->d->v[i][j] = 0.0;
- }
- }
-}
-
-static int ebur128_init_channel_map(FFEBUR128State * st)
-{
- size_t i;
- st->d->channel_map =
- (int *) av_malloc_array(st->channels, sizeof(*st->d->channel_map));
- if (!st->d->channel_map)
- return AVERROR(ENOMEM);
- if (st->channels == 4) {
- st->d->channel_map[0] = FF_EBUR128_LEFT;
- st->d->channel_map[1] = FF_EBUR128_RIGHT;
- st->d->channel_map[2] = FF_EBUR128_LEFT_SURROUND;
- st->d->channel_map[3] = FF_EBUR128_RIGHT_SURROUND;
- } else if (st->channels == 5) {
- st->d->channel_map[0] = FF_EBUR128_LEFT;
- st->d->channel_map[1] = FF_EBUR128_RIGHT;
- st->d->channel_map[2] = FF_EBUR128_CENTER;
- st->d->channel_map[3] = FF_EBUR128_LEFT_SURROUND;
- st->d->channel_map[4] = FF_EBUR128_RIGHT_SURROUND;
- } else {
- for (i = 0; i < st->channels; ++i) {
- switch (i) {
- case 0:
- st->d->channel_map[i] = FF_EBUR128_LEFT;
- break;
- case 1:
- st->d->channel_map[i] = FF_EBUR128_RIGHT;
- break;
- case 2:
- st->d->channel_map[i] = FF_EBUR128_CENTER;
- break;
- case 3:
- st->d->channel_map[i] = FF_EBUR128_UNUSED;
- break;
- case 4:
- st->d->channel_map[i] = FF_EBUR128_LEFT_SURROUND;
- break;
- case 5:
- st->d->channel_map[i] = FF_EBUR128_RIGHT_SURROUND;
- break;
- default:
- st->d->channel_map[i] = FF_EBUR128_UNUSED;
- break;
- }
- }
- }
- return 0;
-}
-
-static inline void init_histogram(void)
-{
- int i;
- /* initialize static constants */
- histogram_energy_boundaries[0] = pow(10.0, (-70.0 + 0.691) / 10.0);
- for (i = 0; i < 1000; ++i) {
- histogram_energies[i] =
- pow(10.0, ((double) i / 10.0 - 69.95 + 0.691) / 10.0);
- }
- for (i = 1; i < 1001; ++i) {
- histogram_energy_boundaries[i] =
- pow(10.0, ((double) i / 10.0 - 70.0 + 0.691) / 10.0);
- }
-}
-
-FFEBUR128State *ff_ebur128_init(unsigned int channels,
- unsigned long samplerate,
- unsigned long window, int mode)
-{
- int errcode;
- FFEBUR128State *st;
-
- st = (FFEBUR128State *) av_malloc(sizeof(*st));
- CHECK_ERROR(!st, 0, exit)
- st->d = (struct FFEBUR128StateInternal *)
- av_malloc(sizeof(*st->d));
- CHECK_ERROR(!st->d, 0, free_state)
- st->channels = channels;
- errcode = ebur128_init_channel_map(st);
- CHECK_ERROR(errcode, 0, free_internal)
-
- st->d->sample_peak =
- (double *) av_calloc(channels, sizeof(*st->d->sample_peak));
- CHECK_ERROR(!st->d->sample_peak, 0, free_channel_map)
-
- st->samplerate = samplerate;
- st->d->samples_in_100ms = (st->samplerate + 5) / 10;
- st->mode = mode;
- if ((mode & FF_EBUR128_MODE_S) == FF_EBUR128_MODE_S) {
- st->d->window = FFMAX(window, 3000);
- } else if ((mode & FF_EBUR128_MODE_M) == FF_EBUR128_MODE_M) {
- st->d->window = FFMAX(window, 400);
- } else {
- goto free_sample_peak;
- }
- st->d->audio_data_frames = st->samplerate * st->d->window / 1000;
- if (st->d->audio_data_frames % st->d->samples_in_100ms) {
- /* round up to multiple of samples_in_100ms */
- st->d->audio_data_frames = st->d->audio_data_frames
- + st->d->samples_in_100ms
- - (st->d->audio_data_frames % st->d->samples_in_100ms);
- }
- st->d->audio_data =
- (double *) av_calloc(st->d->audio_data_frames,
- st->channels * sizeof(*st->d->audio_data));
- CHECK_ERROR(!st->d->audio_data, 0, free_sample_peak)
-
- ebur128_init_filter(st);
-
- st->d->block_energy_histogram =
- av_mallocz(1000 * sizeof(*st->d->block_energy_histogram));
- CHECK_ERROR(!st->d->block_energy_histogram, 0, free_audio_data)
- st->d->short_term_block_energy_histogram =
- av_mallocz(1000 * sizeof(*st->d->short_term_block_energy_histogram));
- CHECK_ERROR(!st->d->short_term_block_energy_histogram, 0,
- free_block_energy_histogram)
- st->d->short_term_frame_counter = 0;
-
- /* the first block needs 400ms of audio data */
- st->d->needed_frames = st->d->samples_in_100ms * 4;
- /* start at the beginning of the buffer */
- st->d->audio_data_index = 0;
-
- if (ff_thread_once(&histogram_init, &init_histogram) != 0)
- goto free_short_term_block_energy_histogram;
-
- st->d->data_ptrs = av_malloc_array(channels, sizeof(*st->d->data_ptrs));
- CHECK_ERROR(!st->d->data_ptrs, 0,
- free_short_term_block_energy_histogram);
-
- return st;
-
-free_short_term_block_energy_histogram:
- av_free(st->d->short_term_block_energy_histogram);
-free_block_energy_histogram:
- av_free(st->d->block_energy_histogram);
-free_audio_data:
- av_free(st->d->audio_data);
-free_sample_peak:
- av_free(st->d->sample_peak);
-free_channel_map:
- av_free(st->d->channel_map);
-free_internal:
- av_free(st->d);
-free_state:
- av_free(st);
-exit:
- return NULL;
-}
-
-void ff_ebur128_destroy(FFEBUR128State ** st)
-{
- av_free((*st)->d->block_energy_histogram);
- av_free((*st)->d->short_term_block_energy_histogram);
- av_free((*st)->d->audio_data);
- av_free((*st)->d->channel_map);
- av_free((*st)->d->sample_peak);
- av_free((*st)->d->data_ptrs);
- av_free((*st)->d);
- av_free(*st);
- *st = NULL;
-}
-
-#define EBUR128_FILTER(type, scaling_factor) \
-static void ebur128_filter_##type(FFEBUR128State* st, const type** srcs, \
- size_t src_index, size_t frames, \
- int stride) { \
- double* audio_data = st->d->audio_data + st->d->audio_data_index; \
- size_t i, c; \
- \
- if ((st->mode & FF_EBUR128_MODE_SAMPLE_PEAK) == FF_EBUR128_MODE_SAMPLE_PEAK) { \
- for (c = 0; c < st->channels; ++c) { \
- double max = 0.0; \
- for (i = 0; i < frames; ++i) { \
- type v = srcs[c][src_index + i * stride]; \
- if (v > max) { \
- max = v; \
- } else if (-v > max) { \
- max = -1.0 * v; \
- } \
- } \
- max /= scaling_factor; \
- if (max > st->d->sample_peak[c]) st->d->sample_peak[c] = max; \
- } \
- } \
- for (c = 0; c < st->channels; ++c) { \
- int ci = st->d->channel_map[c] - 1; \
- if (ci < 0) continue; \
- else if (ci == FF_EBUR128_DUAL_MONO - 1) ci = 0; /*dual mono */ \
- for (i = 0; i < frames; ++i) { \
- st->d->v[ci][0] = (double) (srcs[c][src_index + i * stride] / scaling_factor) \
- - st->d->a[1] * st->d->v[ci][1] \
- - st->d->a[2] * st->d->v[ci][2] \
- - st->d->a[3] * st->d->v[ci][3] \
- - st->d->a[4] * st->d->v[ci][4]; \
- audio_data[i * st->channels + c] = \
- st->d->b[0] * st->d->v[ci][0] \
- + st->d->b[1] * st->d->v[ci][1] \
- + st->d->b[2] * st->d->v[ci][2] \
- + st->d->b[3] * st->d->v[ci][3] \
- + st->d->b[4] * st->d->v[ci][4]; \
- st->d->v[ci][4] = st->d->v[ci][3]; \
- st->d->v[ci][3] = st->d->v[ci][2]; \
- st->d->v[ci][2] = st->d->v[ci][1]; \
- st->d->v[ci][1] = st->d->v[ci][0]; \
- } \
- st->d->v[ci][4] = fabs(st->d->v[ci][4]) < DBL_MIN ? 0.0 : st->d->v[ci][4]; \
- st->d->v[ci][3] = fabs(st->d->v[ci][3]) < DBL_MIN ? 0.0 : st->d->v[ci][3]; \
- st->d->v[ci][2] = fabs(st->d->v[ci][2]) < DBL_MIN ? 0.0 : st->d->v[ci][2]; \
- st->d->v[ci][1] = fabs(st->d->v[ci][1]) < DBL_MIN ? 0.0 : st->d->v[ci][1]; \
- } \
-}
-EBUR128_FILTER(double, 1.0)
-
-static double ebur128_energy_to_loudness(double energy)
-{
- return 10 * log10(energy) - 0.691;
-}
-
-static size_t find_histogram_index(double energy)
-{
- size_t index_min = 0;
- size_t index_max = 1000;
- size_t index_mid;
-
- do {
- index_mid = (index_min + index_max) / 2;
- if (energy >= histogram_energy_boundaries[index_mid]) {
- index_min = index_mid;
- } else {
- index_max = index_mid;
- }
- } while (index_max - index_min != 1);
-
- return index_min;
-}
-
-static void ebur128_calc_gating_block(FFEBUR128State * st,
- size_t frames_per_block,
- double *optional_output)
-{
- size_t i, c;
- double sum = 0.0;
- double channel_sum;
- for (c = 0; c < st->channels; ++c) {
- if (st->d->channel_map[c] == FF_EBUR128_UNUSED)
- continue;
- channel_sum = 0.0;
- if (st->d->audio_data_index < frames_per_block * st->channels) {
- for (i = 0; i < st->d->audio_data_index / st->channels; ++i) {
- channel_sum += st->d->audio_data[i * st->channels + c] *
- st->d->audio_data[i * st->channels + c];
- }
- for (i = st->d->audio_data_frames -
- (frames_per_block -
- st->d->audio_data_index / st->channels);
- i < st->d->audio_data_frames; ++i) {
- channel_sum += st->d->audio_data[i * st->channels + c] *
- st->d->audio_data[i * st->channels + c];
- }
- } else {
- for (i =
- st->d->audio_data_index / st->channels - frames_per_block;
- i < st->d->audio_data_index / st->channels; ++i) {
- channel_sum +=
- st->d->audio_data[i * st->channels +
- c] * st->d->audio_data[i *
- st->channels +
- c];
- }
- }
- if (st->d->channel_map[c] == FF_EBUR128_Mp110 ||
- st->d->channel_map[c] == FF_EBUR128_Mm110 ||
- st->d->channel_map[c] == FF_EBUR128_Mp060 ||
- st->d->channel_map[c] == FF_EBUR128_Mm060 ||
- st->d->channel_map[c] == FF_EBUR128_Mp090 ||
- st->d->channel_map[c] == FF_EBUR128_Mm090) {
- channel_sum *= 1.41;
- } else if (st->d->channel_map[c] == FF_EBUR128_DUAL_MONO) {
- channel_sum *= 2.0;
- }
- sum += channel_sum;
- }
- sum /= (double) frames_per_block;
- if (optional_output) {
- *optional_output = sum;
- } else if (sum >= histogram_energy_boundaries[0]) {
- ++st->d->block_energy_histogram[find_histogram_index(sum)];
- }
-}
-
-int ff_ebur128_set_channel(FFEBUR128State * st,
- unsigned int channel_number, int value)
-{
- if (channel_number >= st->channels) {
- return 1;
- }
- if (value == FF_EBUR128_DUAL_MONO &&
- (st->channels != 1 || channel_number != 0)) {
- return 1;
- }
- st->d->channel_map[channel_number] = value;
- return 0;
-}
-
-static int ebur128_energy_shortterm(FFEBUR128State * st, double *out);
-#define EBUR128_ADD_FRAMES_PLANAR(type) \
-static void ebur128_add_frames_planar_##type(FFEBUR128State* st, const type** srcs, \
- size_t frames, int stride) { \
- size_t src_index = 0; \
- while (frames > 0) { \
- if (frames >= st->d->needed_frames) { \
- ebur128_filter_##type(st, srcs, src_index, st->d->needed_frames, stride); \
- src_index += st->d->needed_frames * stride; \
- frames -= st->d->needed_frames; \
- st->d->audio_data_index += st->d->needed_frames * st->channels; \
- /* calculate the new gating block */ \
- if ((st->mode & FF_EBUR128_MODE_I) == FF_EBUR128_MODE_I) { \
- ebur128_calc_gating_block(st, st->d->samples_in_100ms * 4, NULL); \
- } \
- if ((st->mode & FF_EBUR128_MODE_LRA) == FF_EBUR128_MODE_LRA) { \
- st->d->short_term_frame_counter += st->d->needed_frames; \
- if (st->d->short_term_frame_counter == st->d->samples_in_100ms * 30) { \
- double st_energy; \
- ebur128_energy_shortterm(st, &st_energy); \
- if (st_energy >= histogram_energy_boundaries[0]) { \
- ++st->d->short_term_block_energy_histogram[ \
- find_histogram_index(st_energy)]; \
- } \
- st->d->short_term_frame_counter = st->d->samples_in_100ms * 20; \
- } \
- } \
- /* 100ms are needed for all blocks besides the first one */ \
- st->d->needed_frames = st->d->samples_in_100ms; \
- /* reset audio_data_index when buffer full */ \
- if (st->d->audio_data_index == st->d->audio_data_frames * st->channels) { \
- st->d->audio_data_index = 0; \
- } \
- } else { \
- ebur128_filter_##type(st, srcs, src_index, frames, stride); \
- st->d->audio_data_index += frames * st->channels; \
- if ((st->mode & FF_EBUR128_MODE_LRA) == FF_EBUR128_MODE_LRA) { \
- st->d->short_term_frame_counter += frames; \
- } \
- st->d->needed_frames -= frames; \
- frames = 0; \
- } \
- } \
-}
-EBUR128_ADD_FRAMES_PLANAR(double)
-#define FF_EBUR128_ADD_FRAMES(type) \
-void ff_ebur128_add_frames_##type(FFEBUR128State* st, const type* src, \
- size_t frames) { \
- int i; \
- const type **buf = (const type**)st->d->data_ptrs; \
- for (i = 0; i < st->channels; i++) \
- buf[i] = src + i; \
- ebur128_add_frames_planar_##type(st, buf, frames, st->channels); \
-}
-FF_EBUR128_ADD_FRAMES(double)
-
-static int ebur128_calc_relative_threshold(FFEBUR128State **sts, size_t size,
- double *relative_threshold)
-{
- size_t i, j;
- int above_thresh_counter = 0;
- *relative_threshold = 0.0;
-
- for (i = 0; i < size; i++) {
- unsigned long *block_energy_histogram = sts[i]->d->block_energy_histogram;
- for (j = 0; j < 1000; ++j) {
- *relative_threshold += block_energy_histogram[j] * histogram_energies[j];
- above_thresh_counter += block_energy_histogram[j];
- }
- }
-
- if (above_thresh_counter != 0) {
- *relative_threshold /= (double)above_thresh_counter;
- *relative_threshold *= RELATIVE_GATE_FACTOR;
- }
-
- return above_thresh_counter;
-}
-
-static int ebur128_gated_loudness(FFEBUR128State ** sts, size_t size,
- double *out)
-{
- double gated_loudness = 0.0;
- double relative_threshold;
- size_t above_thresh_counter;
- size_t i, j, start_index;
-
- for (i = 0; i < size; i++)
- if ((sts[i]->mode & FF_EBUR128_MODE_I) != FF_EBUR128_MODE_I)
- return AVERROR(EINVAL);
-
- if (!ebur128_calc_relative_threshold(sts, size, &relative_threshold)) {
- *out = -HUGE_VAL;
- return 0;
- }
-
- above_thresh_counter = 0;
- if (relative_threshold < histogram_energy_boundaries[0]) {
- start_index = 0;
- } else {
- start_index = find_histogram_index(relative_threshold);
- if (relative_threshold > histogram_energies[start_index]) {
- ++start_index;
- }
- }
- for (i = 0; i < size; i++) {
- for (j = start_index; j < 1000; ++j) {
- gated_loudness += sts[i]->d->block_energy_histogram[j] *
- histogram_energies[j];
- above_thresh_counter += sts[i]->d->block_energy_histogram[j];
- }
- }
- if (!above_thresh_counter) {
- *out = -HUGE_VAL;
- return 0;
- }
- gated_loudness /= (double) above_thresh_counter;
- *out = ebur128_energy_to_loudness(gated_loudness);
- return 0;
-}
-
-int ff_ebur128_relative_threshold(FFEBUR128State * st, double *out)
-{
- double relative_threshold;
-
- if ((st->mode & FF_EBUR128_MODE_I) != FF_EBUR128_MODE_I)
- return AVERROR(EINVAL);
-
- if (!ebur128_calc_relative_threshold(&st, 1, &relative_threshold)) {
- *out = -70.0;
- return 0;
- }
-
- *out = ebur128_energy_to_loudness(relative_threshold);
- return 0;
-}
-
-int ff_ebur128_loudness_global(FFEBUR128State * st, double *out)
-{
- return ebur128_gated_loudness(&st, 1, out);
-}
-
-static int ebur128_energy_in_interval(FFEBUR128State * st,
- size_t interval_frames, double *out)
-{
- if (interval_frames > st->d->audio_data_frames) {
- return AVERROR(EINVAL);
- }
- ebur128_calc_gating_block(st, interval_frames, out);
- return 0;
-}
-
-static int ebur128_energy_shortterm(FFEBUR128State * st, double *out)
-{
- return ebur128_energy_in_interval(st, st->d->samples_in_100ms * 30,
- out);
-}
-
-int ff_ebur128_loudness_shortterm(FFEBUR128State * st, double *out)
-{
- double energy;
- int error = ebur128_energy_shortterm(st, &energy);
- if (error) {
- return error;
- } else if (energy <= 0.0) {
- *out = -HUGE_VAL;
- return 0;
- }
- *out = ebur128_energy_to_loudness(energy);
- return 0;
-}
-
-/* EBU - TECH 3342 */
-int ff_ebur128_loudness_range_multiple(FFEBUR128State ** sts, size_t size,
- double *out)
-{
- size_t i, j;
- size_t stl_size;
- double stl_power, stl_integrated;
- /* High and low percentile energy */
- double h_en, l_en;
- unsigned long hist[1000] = { 0 };
- size_t percentile_low, percentile_high;
- size_t index;
-
- for (i = 0; i < size; ++i) {
- if (sts[i]) {
- if ((sts[i]->mode & FF_EBUR128_MODE_LRA) !=
- FF_EBUR128_MODE_LRA) {
- return AVERROR(EINVAL);
- }
- }
- }
-
- stl_size = 0;
- stl_power = 0.0;
- for (i = 0; i < size; ++i) {
- if (!sts[i])
- continue;
- for (j = 0; j < 1000; ++j) {
- hist[j] += sts[i]->d->short_term_block_energy_histogram[j];
- stl_size += sts[i]->d->short_term_block_energy_histogram[j];
- stl_power += sts[i]->d->short_term_block_energy_histogram[j]
- * histogram_energies[j];
- }
- }
- if (!stl_size) {
- *out = 0.0;
- return 0;
- }
-
- stl_power /= stl_size;
- stl_integrated = MINUS_20DB * stl_power;
-
- if (stl_integrated < histogram_energy_boundaries[0]) {
- index = 0;
- } else {
- index = find_histogram_index(stl_integrated);
- if (stl_integrated > histogram_energies[index]) {
- ++index;
- }
- }
- stl_size = 0;
- for (j = index; j < 1000; ++j) {
- stl_size += hist[j];
- }
- if (!stl_size) {
- *out = 0.0;
- return 0;
- }
-
- percentile_low = (size_t) ((stl_size - 1) * 0.1 + 0.5);
- percentile_high = (size_t) ((stl_size - 1) * 0.95 + 0.5);
-
- stl_size = 0;
- j = index;
- while (stl_size <= percentile_low) {
- stl_size += hist[j++];
- }
- l_en = histogram_energies[j - 1];
- while (stl_size <= percentile_high) {
- stl_size += hist[j++];
- }
- h_en = histogram_energies[j - 1];
- *out =
- ebur128_energy_to_loudness(h_en) -
- ebur128_energy_to_loudness(l_en);
- return 0;
-}
-
-int ff_ebur128_loudness_range(FFEBUR128State * st, double *out)
-{
- return ff_ebur128_loudness_range_multiple(&st, 1, out);
-}
-
-int ff_ebur128_sample_peak(FFEBUR128State * st,
- unsigned int channel_number, double *out)
-{
- if ((st->mode & FF_EBUR128_MODE_SAMPLE_PEAK) !=
- FF_EBUR128_MODE_SAMPLE_PEAK) {
- return AVERROR(EINVAL);
- } else if (channel_number >= st->channels) {
- return AVERROR(EINVAL);
- }
- *out = st->d->sample_peak[channel_number];
- return 0;
-}
diff --git a/libavfilter/ebur128.h b/libavfilter/ebur128.h
deleted file mode 100644
index 8e7385e044..0000000000
--- a/libavfilter/ebur128.h
+++ /dev/null
@@ -1,229 +0,0 @@
-/*
- * Copyright (c) 2011 Jan Kokemüller
- *
- * 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
- *
- * This file is based on libebur128 which is available at
- * https://github.com/jiixyj/libebur128/
- *
-*/
-
-#ifndef AVFILTER_EBUR128_H
-#define AVFILTER_EBUR128_H
-
-/** \file ebur128.h
- * \brief libebur128 - a library for loudness measurement according to
- * the EBU R128 standard.
- */
-
-#include <stddef.h> /* for size_t */
-
-/** \enum channel
- * Use these values when setting the channel map with ebur128_set_channel().
- * See definitions in ITU R-REC-BS 1770-4
- */
-enum channel {
- FF_EBUR128_UNUSED = 0, /**< unused channel (for example LFE channel) */
- FF_EBUR128_LEFT,
- FF_EBUR128_Mp030 = 1, /**< itu M+030 */
- FF_EBUR128_RIGHT,
- FF_EBUR128_Mm030 = 2, /**< itu M-030 */
- FF_EBUR128_CENTER,
- FF_EBUR128_Mp000 = 3, /**< itu M+000 */
- FF_EBUR128_LEFT_SURROUND,
- FF_EBUR128_Mp110 = 4, /**< itu M+110 */
- FF_EBUR128_RIGHT_SURROUND,
- FF_EBUR128_Mm110 = 5, /**< itu M-110 */
- FF_EBUR128_DUAL_MONO, /**< a channel that is counted twice */
- FF_EBUR128_MpSC, /**< itu M+SC */
- FF_EBUR128_MmSC, /**< itu M-SC */
- FF_EBUR128_Mp060, /**< itu M+060 */
- FF_EBUR128_Mm060, /**< itu M-060 */
- FF_EBUR128_Mp090, /**< itu M+090 */
- FF_EBUR128_Mm090, /**< itu M-090 */
- FF_EBUR128_Mp135, /**< itu M+135 */
- FF_EBUR128_Mm135, /**< itu M-135 */
- FF_EBUR128_Mp180, /**< itu M+180 */
- FF_EBUR128_Up000, /**< itu U+000 */
- FF_EBUR128_Up030, /**< itu U+030 */
- FF_EBUR128_Um030, /**< itu U-030 */
- FF_EBUR128_Up045, /**< itu U+045 */
- FF_EBUR128_Um045, /**< itu U-030 */
- FF_EBUR128_Up090, /**< itu U+090 */
- FF_EBUR128_Um090, /**< itu U-090 */
- FF_EBUR128_Up110, /**< itu U+110 */
- FF_EBUR128_Um110, /**< itu U-110 */
- FF_EBUR128_Up135, /**< itu U+135 */
- FF_EBUR128_Um135, /**< itu U-135 */
- FF_EBUR128_Up180, /**< itu U+180 */
- FF_EBUR128_Tp000, /**< itu T+000 */
- FF_EBUR128_Bp000, /**< itu B+000 */
- FF_EBUR128_Bp045, /**< itu B+045 */
- FF_EBUR128_Bm045 /**< itu B-045 */
-};
-
-/** \enum mode
- * Use these values in ebur128_init (or'ed). Try to use the lowest possible
- * modes that suit your needs, as performance will be better.
- */
-enum mode {
- /** can resurrrect and call ff_ebur128_loudness_momentary */
- FF_EBUR128_MODE_M = (1 << 0),
- /** can call ff_ebur128_loudness_shortterm */
- FF_EBUR128_MODE_S = (1 << 1) | FF_EBUR128_MODE_M,
- /** can call ff_ebur128_loudness_global_* and ff_ebur128_relative_threshold */
- FF_EBUR128_MODE_I = (1 << 2) | FF_EBUR128_MODE_M,
- /** can call ff_ebur128_loudness_range */
- FF_EBUR128_MODE_LRA = (1 << 3) | FF_EBUR128_MODE_S,
- /** can call ff_ebur128_sample_peak */
- FF_EBUR128_MODE_SAMPLE_PEAK = (1 << 4) | FF_EBUR128_MODE_M,
-};
-
-/** forward declaration of FFEBUR128StateInternal */
-struct FFEBUR128StateInternal;
-
-/** \brief Contains information about the state of a loudness measurement.
- *
- * You should not need to modify this struct directly.
- */
-typedef struct FFEBUR128State {
- int mode; /**< The current mode. */
- unsigned int channels; /**< The number of channels. */
- unsigned long samplerate; /**< The sample rate. */
- struct FFEBUR128StateInternal *d; /**< Internal state. */
-} FFEBUR128State;
-
-/** \brief Initialize library state.
- *
- * @param channels the number of channels.
- * @param samplerate the sample rate.
- * @param window set the maximum window size in ms, set to 0 for auto.
- * @param mode see the mode enum for possible values.
- * @return an initialized library state.
- */
-FFEBUR128State *ff_ebur128_init(unsigned int channels,
- unsigned long samplerate,
- unsigned long window, int mode);
-
-/** \brief Destroy library state.
- *
- * @param st pointer to a library state.
- */
-void ff_ebur128_destroy(FFEBUR128State ** st);
-
-/** \brief Set channel type.
- *
- * The default is:
- * - 0 -> FF_EBUR128_LEFT
- * - 1 -> FF_EBUR128_RIGHT
- * - 2 -> FF_EBUR128_CENTER
- * - 3 -> FF_EBUR128_UNUSED
- * - 4 -> FF_EBUR128_LEFT_SURROUND
- * - 5 -> FF_EBUR128_RIGHT_SURROUND
- *
- * @param st library state.
- * @param channel_number zero based channel index.
- * @param value channel type from the "channel" enum.
- * @return
- * - 0 on success.
- * - AVERROR(EINVAL) if invalid channel index.
- */
-int ff_ebur128_set_channel(FFEBUR128State * st,
- unsigned int channel_number, int value);
-
-/** \brief Add frames to be processed.
- *
- * @param st library state.
- * @param src array of source frames. Channels must be interleaved.
- * @param frames number of frames. Not number of samples!
- */
-void ff_ebur128_add_frames_double(FFEBUR128State * st,
- const double *src, size_t frames);
-
-/** \brief Get global integrated loudness in LUFS.
- *
- * @param st library state.
- * @param out integrated loudness in LUFS. -HUGE_VAL if result is negative
- * infinity.
- * @return
- * - 0 on success.
- * - AVERROR(EINVAL) if mode "FF_EBUR128_MODE_I" has not been set.
- */
-int ff_ebur128_loudness_global(FFEBUR128State * st, double *out);
-
-/** \brief Get short-term loudness (last 3s) in LUFS.
- *
- * @param st library state.
- * @param out short-term loudness in LUFS. -HUGE_VAL if result is negative
- * infinity.
- * @return
- * - 0 on success.
- * - AVERROR(EINVAL) if mode "FF_EBUR128_MODE_S" has not been set.
- */
-int ff_ebur128_loudness_shortterm(FFEBUR128State * st, double *out);
-
-/** \brief Get loudness range (LRA) of programme in LU.
- *
- * Calculates loudness range according to EBU 3342.
- *
- * @param st library state.
- * @param out loudness range (LRA) in LU. Will not be changed in case of
- * error. AVERROR(EINVAL) will be returned in this case.
- * @return
- * - 0 on success.
- * - AVERROR(EINVAL) if mode "FF_EBUR128_MODE_LRA" has not been set.
- */
-int ff_ebur128_loudness_range(FFEBUR128State * st, double *out);
-/** \brief Get loudness range (LRA) in LU across multiple instances.
- *
- * Calculates loudness range according to EBU 3342.
- *
- * @param sts array of library states.
- * @param size length of sts
- * @param out loudness range (LRA) in LU. Will not be changed in case of
- * error. AVERROR(EINVAL) will be returned in this case.
- * @return
- * - 0 on success.
- * - AVERROR(EINVAL) if mode "FF_EBUR128_MODE_LRA" has not been set.
- */
-int ff_ebur128_loudness_range_multiple(FFEBUR128State ** sts,
- size_t size, double *out);
-
-/** \brief Get maximum sample peak of selected channel in float format.
- *
- * @param st library state
- * @param channel_number channel to analyse
- * @param out maximum sample peak in float format (1.0 is 0 dBFS)
- * @return
- * - 0 on success.
- * - AVERROR(EINVAL) if mode "FF_EBUR128_MODE_SAMPLE_PEAK" has not been set.
- * - AVERROR(EINVAL) if invalid channel index.
- */
-int ff_ebur128_sample_peak(FFEBUR128State * st,
- unsigned int channel_number, double *out);
-
-/** \brief Get relative threshold in LUFS.
- *
- * @param st library state
- * @param out relative threshold in LUFS.
- * @return
- * - 0 on success.
- * - AVERROR(EINVAL) if mode "FF_EBUR128_MODE_I" has not been set.
- */
-int ff_ebur128_relative_threshold(FFEBUR128State * st, double *out);
-
-#endif /* AVFILTER_EBUR128_H */
diff --git a/libavfilter/f_ebur128.c b/libavfilter/f_ebur128.c
index a921602b44..4508d3039c 100644
--- a/libavfilter/f_ebur128.c
+++ b/libavfilter/f_ebur128.c
@@ -75,6 +75,13 @@ struct integrator {
struct hist_entry *histogram; ///< histogram of the powers, used to compute LRA and I
};
+enum PrintFormat {
+ NONE,
+ JSON,
+ SUMMARY,
+ PF_NB
+};
+
struct rect { int x, y, w, h; };
typedef struct EBUR128Context {
@@ -389,11 +396,8 @@ static int config_video_output(AVFilterLink *outlink)
return 0;
}
-static int config_audio_input(AVFilterLink *inlink)
+static int config_audio_in(AVFilterLink *inlink, EBUR128Context *ebur128)
{
- AVFilterContext *ctx = inlink->dst;
- EBUR128Context *ebur128 = ctx->priv;
-
/* Unofficial reversed parametrization of PRE
* and RLB from 48kHz */
@@ -434,11 +438,16 @@ static int config_audio_input(AVFilterLink *inlink)
return 0;
}
-static int config_audio_output(AVFilterLink *outlink)
+static int config_audio_input(AVFilterLink *inlink)
{
- int i;
- AVFilterContext *ctx = outlink->src;
+ AVFilterContext *ctx = inlink->dst;
EBUR128Context *ebur128 = ctx->priv;
+ return config_audio_in(inlink, ebur128);
+}
+
+static int config_audio_out(AVFilterLink *outlink, EBUR128Context *ebur128)
+{
+ int i;
const int nb_channels = outlink->ch_layout.nb_channels;
#define BACK_MASK (AV_CH_BACK_LEFT |AV_CH_BACK_CENTER |AV_CH_BACK_RIGHT| \
@@ -523,6 +532,13 @@ static int config_audio_output(AVFilterLink *outlink)
return 0;
}
+static int config_audio_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ EBUR128Context *ebur128 = ctx->priv;
+ return config_audio_out(outlink, ebur128);
+}
+
#define ENERGY(loudness) (ff_exp10(((loudness) + 0.691) / 10.))
#define LOUDNESS(energy) (-0.691 + 10 * log10(energy))
#define DBFS(energy) (20 * log10(energy))
@@ -541,9 +557,8 @@ static struct hist_entry *get_histogram(void)
return h;
}
-static av_cold int init(AVFilterContext *ctx)
+static av_cold int init_ebur128(AVFilterContext *ctx, EBUR128Context *ebur128)
{
- EBUR128Context *ebur128 = ctx->priv;
AVFilterPad pad;
int ret;
@@ -574,6 +589,9 @@ static av_cold int init(AVFilterContext *ctx)
ebur128->integrated_loudness = ABS_THRES;
ebur128->loudness_range = 0;
+ if (strcmp(ctx->filter->name, "ebur128"))
+ return 0;
+
/* insert output pads */
if (ebur128->do_video) {
pad = (AVFilterPad){
@@ -600,6 +618,12 @@ static av_cold int init(AVFilterContext *ctx)
return 0;
}
+static av_cold int init(AVFilterContext *ctx)
+{
+ EBUR128Context *ebur128 = ctx->priv;
+ return init_ebur128(ctx, ebur128);
+}
+
#define HIST_POS(power) (int)(((power) - ABS_THRES) * HIST_GRAIN)
/* loudness and power should be set such as loudness = -0.691 +
@@ -627,27 +651,21 @@ static int gate_update(struct integrator *integ, double power,
return gate_hist_pos;
}
-static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
+static int true_peaks_ebur128(EBUR128Context *ebur128, const uint8_t **samples,
+ int nb_samples)
{
- int i, ch, idx_insample, ret;
- AVFilterContext *ctx = inlink->dst;
- EBUR128Context *ebur128 = ctx->priv;
- const int nb_channels = ebur128->nb_channels;
- const int nb_samples = insamples->nb_samples;
- const double *samples = (double *)insamples->data[0];
- AVFrame *pic;
-
#if CONFIG_SWRESAMPLE
if (ebur128->peak_mode & PEAK_MODE_TRUE_PEAKS && ebur128->idx_insample == 0) {
const double *swr_samples = ebur128->swr_buf;
+ const int nb_channels = ebur128->nb_channels;
int ret = swr_convert(ebur128->swr_ctx, (uint8_t**)&ebur128->swr_buf, 19200,
- (const uint8_t **)insamples->data, nb_samples);
+ samples, nb_samples);
if (ret < 0)
return ret;
- for (ch = 0; ch < nb_channels; ch++)
+ for (int ch = 0; ch < nb_channels; ch++)
ebur128->true_peaks_per_frame[ch] = 0.0;
- for (idx_insample = 0; idx_insample < ret; idx_insample++) {
- for (ch = 0; ch < nb_channels; ch++) {
+ for (int idx_insample = 0; idx_insample < ret; idx_insample++) {
+ for (int ch = 0; ch < nb_channels; ch++) {
ebur128->true_peaks[ch] = FFMAX(ebur128->true_peaks[ch], fabs(*swr_samples));
ebur128->true_peaks_per_frame[ch] = FFMAX(ebur128->true_peaks_per_frame[ch],
fabs(*swr_samples));
@@ -656,10 +674,14 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
}
}
#endif
+ return 0;
+}
- for (idx_insample = ebur128->idx_insample; idx_insample < nb_samples; idx_insample++) {
- const int bin_id_400 = ebur128->i400.cache_pos;
- const int bin_id_3000 = ebur128->i3000.cache_pos;
+static void process_ebur128(EBUR128Context *ebur128, const double *samples)
+{
+ const int nb_channels = ebur128->nb_channels;
+ const int bin_id_400 = ebur128->i400.cache_pos;
+ const int bin_id_3000 = ebur128->i3000.cache_pos;
#define MOVE_TO_NEXT_CACHED_ENTRY(time) do { \
ebur128->i##time.cache_pos++; \
@@ -670,47 +692,47 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
} \
} while (0)
- MOVE_TO_NEXT_CACHED_ENTRY(400);
- MOVE_TO_NEXT_CACHED_ENTRY(3000);
+ MOVE_TO_NEXT_CACHED_ENTRY(400);
+ MOVE_TO_NEXT_CACHED_ENTRY(3000);
- for (ch = 0; ch < nb_channels; ch++) {
- double bin;
+ for (int ch = 0; ch < nb_channels; ch++) {
+ double bin;
- if (ebur128->peak_mode & PEAK_MODE_SAMPLES_PEAKS)
- ebur128->sample_peaks[ch] = FFMAX(ebur128->sample_peaks[ch], fabs(samples[idx_insample * nb_channels + ch]));
+ if (ebur128->peak_mode & PEAK_MODE_SAMPLES_PEAKS)
+ ebur128->sample_peaks[ch] = FFMAX(ebur128->sample_peaks[ch], fabs(samples[ch]));
- ebur128->x[ch * 3] = samples[idx_insample * nb_channels + ch]; // set X[i]
+ ebur128->x[ch * 3] = samples[ch]; // set X[i]
- if (!ebur128->ch_weighting[ch])
- continue;
+ 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, NUM, DEN) 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]*NUM[0] + src[1]*NUM[1] + src[2]*NUM[2] \
- - dst[1]*DEN[1] - dst[2]*DEN[2]; \
+ /* 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, NUM, DEN) 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]*NUM[0] + src[1]*NUM[1] + src[2]*NUM[2] \
+ - dst[1]*DEN[1] - dst[2]*DEN[2]; \
} while (0)
- // TODO: merge both filters in one?
- FILTER(y, x, ebur128->pre_b, ebur128->pre_a); // 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, ebur128->rlb_b, ebur128->rlb_a); // apply RLB-filter
+ // TODO: merge both filters in one?
+ FILTER(y, x, ebur128->pre_b, ebur128->pre_a); // 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, ebur128->rlb_b, ebur128->rlb_a); // apply RLB-filter
- bin = ebur128->z[ch * 3] * ebur128->z[ch * 3];
+ 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];
+ /* 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;
- }
+ /* override old cache entry with the new value */
+ ebur128->i400.cache [ch][bin_id_400 ] = bin;
+ ebur128->i3000.cache[ch][bin_id_3000] = bin;
+ }
#define FIND_PEAK(global, sp, ptype) do { \
int ch; \
@@ -723,110 +745,141 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
} \
} while (0)
- FIND_PEAK(ebur128->sample_peak, ebur128->sample_peaks, SAMPLES);
- FIND_PEAK(ebur128->true_peak, ebur128->true_peaks, TRUE);
+ FIND_PEAK(ebur128->sample_peak, ebur128->sample_peaks, SAMPLES);
+ FIND_PEAK(ebur128->true_peak, ebur128->true_peaks, TRUE);
+}
- /* For integrated loudness, gating blocks are 400ms long with 75%
- * overlap (see BS.1770-2 p5), so a re-computation is needed each 100ms
- * (4800 samples at 48kHz). */
- if (++ebur128->sample_count == inlink->sample_rate / 10) {
- double loudness_400, loudness_3000;
- double power_400 = 1e-12, power_3000 = 1e-12;
- AVFilterLink *outlink = ctx->outputs[0];
- const int64_t pts = insamples->pts +
- av_rescale_q(idx_insample, (AVRational){ 1, inlink->sample_rate },
- ctx->outputs[ebur128->do_video]->time_base);
+static void ebur128_loudness(AVFilterLink *inlink,
+ EBUR128Context *ebur128,
+ double *l400, double *l3000)
+{
+ const int nb_channels = ebur128->nb_channels;
+ double power_400 = 1e-12, power_3000 = 1e-12;
+ double loudness_400, loudness_3000;
- ebur128->sample_count = 0;
+ ebur128->sample_count = 0;
#define COMPUTE_LOUDNESS(m, time) do { \
if (ebur128->i##time.filled) { \
/* weighting sum of the last <time> ms */ \
- for (ch = 0; ch < nb_channels; ch++) \
+ for (int ch = 0; ch < nb_channels; ch++) \
power_##time += ebur128->ch_weighting[ch] * ebur128->i##time.sum[ch]; \
power_##time /= I##time##_BINS(inlink->sample_rate); \
} \
loudness_##time = LOUDNESS(power_##time); \
} while (0)
- COMPUTE_LOUDNESS(M, 400);
- COMPUTE_LOUDNESS(S, 3000);
+ COMPUTE_LOUDNESS(M, 400);
+ COMPUTE_LOUDNESS(S, 3000);
- /* Integrated loudness */
+ /* 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 = 0.0;
- uint64_t 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 unsigned nb_v = ebur128->i400.histogram[i].count;
- nb_integrated += nb_v;
- integrated_sum += nb_v * ebur128->i400.histogram[i].energy;
- }
- if (nb_integrated) {
- ebur128->integrated_loudness = LOUDNESS(integrated_sum / nb_integrated);
- /* dual-mono correction */
- if (nb_channels == 1 && ebur128->dual_mono) {
- ebur128->integrated_loudness -= ebur128->pan_law;
- }
- }
+ if (loudness_400 >= ABS_THRES) {
+ double integrated_sum = 0.0;
+ uint64_t 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 (int i = gate_hist_pos; i < HIST_SIZE; i++) {
+ const unsigned nb_v = ebur128->i400.histogram[i].count;
+ nb_integrated += nb_v;
+ integrated_sum += nb_v * ebur128->i400.histogram[i].energy;
+ }
+ if (nb_integrated) {
+ ebur128->integrated_loudness = LOUDNESS(integrated_sum / nb_integrated);
+ /* dual-mono correction */
+ if (nb_channels == 1 && ebur128->dual_mono) {
+ ebur128->integrated_loudness -= ebur128->pan_law;
}
+ }
+ }
- /* LRA */
+ /* 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) {
- uint64_t 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.histogram[i].count;
- if (nb_powers) {
- uint64_t n, nb_pow;
-
- /* get lower loudness to consider */
- n = 0;
- nb_pow = LRA_LOWER_PRC * nb_powers * 0.01 + 0.5;
- for (i = gate_hist_pos; i < HIST_SIZE; i++) {
- n += ebur128->i3000.histogram[i].count;
- if (n >= nb_pow) {
- ebur128->lra_low = ebur128->i3000.histogram[i].loudness;
- break;
- }
- }
-
- /* get higher loudness to consider */
- n = nb_powers;
- nb_pow = LRA_HIGHER_PRC * nb_powers * 0.01 + 0.5;
- for (i = HIST_SIZE - 1; i >= 0; i--) {
- n -= FFMIN(n, ebur128->i3000.histogram[i].count);
- if (n < nb_pow) {
- ebur128->lra_high = ebur128->i3000.histogram[i].loudness;
- break;
- }
- }
-
- // XXX: show low & high on the graph?
- ebur128->loudness_range = ebur128->lra_high - ebur128->lra_low;
+ /* XXX: example code in EBU 3342 is ">=" but formula in BS.1770
+ * specs is ">" */
+ if (loudness_3000 >= ABS_THRES) {
+ uint64_t nb_powers = 0;
+ int gate_hist_pos = gate_update(&ebur128->i3000, power_3000,
+ loudness_3000, LRA_GATE_THRES);
+
+ for (int i = gate_hist_pos; i < HIST_SIZE; i++)
+ nb_powers += ebur128->i3000.histogram[i].count;
+ if (nb_powers) {
+ uint64_t n, nb_pow;
+
+ /* get lower loudness to consider */
+ n = 0;
+ nb_pow = LRA_LOWER_PRC * nb_powers * 0.01 + 0.5;
+ for (int i = gate_hist_pos; i < HIST_SIZE; i++) {
+ n += ebur128->i3000.histogram[i].count;
+ if (n >= nb_pow) {
+ ebur128->lra_low = ebur128->i3000.histogram[i].loudness;
+ break;
}
}
- /* dual-mono correction */
- if (nb_channels == 1 && ebur128->dual_mono) {
- loudness_400 -= ebur128->pan_law;
- loudness_3000 -= ebur128->pan_law;
+ /* get higher loudness to consider */
+ n = nb_powers;
+ nb_pow = LRA_HIGHER_PRC * nb_powers * 0.01 + 0.5;
+ for (int i = HIST_SIZE - 1; i >= 0; i--) {
+ n -= FFMIN(n, ebur128->i3000.histogram[i].count);
+ if (n < nb_pow) {
+ ebur128->lra_high = ebur128->i3000.histogram[i].loudness;
+ break;
+ }
}
+ // XXX: show low & high on the graph?
+ ebur128->loudness_range = ebur128->lra_high - ebur128->lra_low;
+ }
+ }
+
+ /* dual-mono correction */
+ if (nb_channels == 1 && ebur128->dual_mono) {
+ loudness_400 -= ebur128->pan_law;
+ loudness_3000 -= ebur128->pan_law;
+ }
+
+ *l400 = loudness_400;
+ *l3000 = loudness_3000;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
+{
+ int idx_insample, ret;
+ AVFilterContext *ctx = inlink->dst;
+ EBUR128Context *ebur128 = ctx->priv;
+ const int nb_channels = ebur128->nb_channels;
+ const int nb_samples = insamples->nb_samples;
+ const double *samples = (const double *)insamples->data[0];
+ AVFrame *pic;
+
+ ret = true_peaks_ebur128(ebur128, (const uint8_t **)insamples->data, nb_samples);
+ if (ret < 0)
+ return ret;
+
+ for (idx_insample = ebur128->idx_insample; idx_insample < nb_samples; idx_insample++) {
+ process_ebur128(ebur128, samples + idx_insample * nb_channels);
+
+ /* For integrated loudness, gating blocks are 400ms long with 75%
+ * overlap (see BS.1770-2 p5), so a re-computation is needed each 100ms
+ * (4800 samples at 48kHz). */
+ if (++ebur128->sample_count == inlink->sample_rate / 10) {
+ double loudness_400, loudness_3000;
+ AVFilterLink *outlink = ctx->outputs[0];
+ const int64_t pts = insamples->pts +
+ av_rescale_q(idx_insample, (AVRational){ 1, inlink->sample_rate },
+ ctx->outputs[ebur128->do_video]->time_base);
+
+ ebur128_loudness(inlink, ebur128, &loudness_400, &loudness_3000);
+
#define LOG_FMT "TARGET:%d LUFS M:%6.1f S:%6.1f I:%6.1f %s LRA:%6.1f LU"
/* push one video frame */
@@ -910,7 +963,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
if (ebur128->peak_mode & PEAK_MODE_ ## ptype ## _PEAKS) { \
double max_peak = 0.0; \
char key[64]; \
- for (ch = 0; ch < nb_channels; ch++) { \
+ for (int ch = 0; ch < nb_channels; ch++) { \
snprintf(key, sizeof(key), \
META_PREFIX AV_STRINGIFY(name) "_peaks_ch%d", ch); \
max_peak = fmax(max_peak, ebur128->name##_peaks[ch]); \
@@ -949,7 +1002,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
#define PRINT_PEAKS(str, sp, ptype) do { \
if (ebur128->peak_mode & PEAK_MODE_ ## ptype ## _PEAKS) { \
av_log(ctx, ebur128->loglevel, " " str ":"); \
- for (ch = 0; ch < nb_channels; ch++) \
+ for (int ch = 0; ch < nb_channels; ch++) \
av_log(ctx, ebur128->loglevel, " %5.1f", DBFS(sp[ch])); \
av_log(ctx, ebur128->loglevel, " dBFS"); \
} \
@@ -1047,6 +1100,35 @@ static int query_formats(AVFilterContext *ctx)
return 0;
}
+static av_cold void uninit_ebur128(AVFilterContext *ctx, EBUR128Context *ebur128)
+{
+ av_freep(&ebur128->y_line_ref);
+ av_freep(&ebur128->x);
+ av_freep(&ebur128->y);
+ av_freep(&ebur128->z);
+ av_freep(&ebur128->ch_weighting);
+ av_freep(&ebur128->true_peaks);
+ av_freep(&ebur128->sample_peaks);
+ av_freep(&ebur128->true_peaks_per_frame);
+ av_freep(&ebur128->i400.sum);
+ av_freep(&ebur128->i3000.sum);
+ av_freep(&ebur128->i400.histogram);
+ av_freep(&ebur128->i3000.histogram);
+ for (int i = 0; i < ebur128->nb_channels; i++) {
+ if (ebur128->i400.cache)
+ av_freep(&ebur128->i400.cache[i]);
+ if (ebur128->i3000.cache)
+ av_freep(&ebur128->i3000.cache[i]);
+ }
+ av_freep(&ebur128->i400.cache);
+ av_freep(&ebur128->i3000.cache);
+ av_frame_free(&ebur128->outpicref);
+#if CONFIG_SWRESAMPLE
+ av_freep(&ebur128->swr_buf);
+ swr_free(&ebur128->swr_ctx);
+#endif
+}
+
static av_cold void uninit(AVFilterContext *ctx)
{
EBUR128Context *ebur128 = ctx->priv;
@@ -1085,31 +1167,7 @@ static av_cold void uninit(AVFilterContext *ctx)
av_log(ctx, AV_LOG_INFO, "\n");
}
- av_freep(&ebur128->y_line_ref);
- av_freep(&ebur128->x);
- av_freep(&ebur128->y);
- av_freep(&ebur128->z);
- av_freep(&ebur128->ch_weighting);
- av_freep(&ebur128->true_peaks);
- av_freep(&ebur128->sample_peaks);
- av_freep(&ebur128->true_peaks_per_frame);
- av_freep(&ebur128->i400.sum);
- av_freep(&ebur128->i3000.sum);
- av_freep(&ebur128->i400.histogram);
- av_freep(&ebur128->i3000.histogram);
- for (int i = 0; i < ebur128->nb_channels; i++) {
- if (ebur128->i400.cache)
- av_freep(&ebur128->i400.cache[i]);
- if (ebur128->i3000.cache)
- av_freep(&ebur128->i3000.cache[i]);
- }
- av_freep(&ebur128->i400.cache);
- av_freep(&ebur128->i3000.cache);
- av_frame_free(&ebur128->outpicref);
-#if CONFIG_SWRESAMPLE
- av_freep(&ebur128->swr_buf);
- swr_free(&ebur128->swr_ctx);
-#endif
+ uninit_ebur128(ctx, ebur128);
}
static const AVFilterPad ebur128_inputs[] = {
@@ -1133,3 +1191,343 @@ const AVFilter ff_af_ebur128 = {
.priv_class = &ebur128_class,
.flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS,
};
+
+typedef struct LoudNormContext {
+ const AVClass *class;
+ double target_i;
+ double target_lra;
+ double target_tp;
+ double measured_i;
+ double measured_lra;
+ double measured_tp;
+ double measured_thresh;
+ double offset;
+ int linear_mode;
+ int dual_mono;
+ enum PrintFormat print_format;
+
+ int nb_channels;
+ int nb_samples;
+
+ AVFrame *insamples;
+
+ EBUR128Context r128_in;
+ EBUR128Context r128_out;
+} LoudNormContext;
+
+static av_cold int loudnorm_init(AVFilterContext *ctx)
+{
+ LoudNormContext *s = ctx->priv;
+ int ret;
+
+ ret = init_ebur128(ctx, &s->r128_in);
+ if (ret < 0)
+ return ret;
+ ret = init_ebur128(ctx, &s->r128_out);
+ if (ret < 0)
+ return ret;
+
+ if (s->linear_mode) {
+ double offset, offset_tp;
+ offset = s->target_i - s->measured_i;
+ offset_tp = s->measured_tp + offset;
+
+ if (s->measured_tp != 99 && s->measured_thresh != -70 && s->measured_lra != 0 && s->measured_i != 0) {
+ if ((offset_tp <= s->target_tp) && (s->measured_lra <= s->target_lra)) {
+ s->offset = pow(10., offset / 20.);
+ }
+ } else {
+ s->linear_mode = 0;
+ }
+ }
+
+ return 0;
+}
+
+static int loudnorm_query_formats(AVFilterContext *ctx)
+{
+ LoudNormContext *s = ctx->priv;
+ static const int input_srate[] = {192000, -1};
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBL,
+ AV_SAMPLE_FMT_NONE
+ };
+ int ret = ff_set_common_all_channel_counts(ctx);
+ if (ret < 0)
+ return ret;
+
+ ret = ff_set_common_formats_from_list(ctx, sample_fmts);
+ if (ret < 0)
+ return ret;
+
+ if (s->linear_mode) {
+ return ff_set_common_all_samplerates(ctx);
+ } else {
+ return ff_set_common_samplerates_from_list(ctx, input_srate);
+ }
+}
+
+static av_cold void loudnorm_uninit(AVFilterContext *ctx)
+{
+ LoudNormContext *s = ctx->priv;
+ EBUR128Context *r128_out = &s->r128_out;
+ EBUR128Context *r128_in = &s->r128_in;
+
+ if (s->nb_channels > 0) {
+ switch (s->print_format) {
+ case NONE:
+ break;
+
+ case JSON:
+ av_log(ctx, AV_LOG_INFO,
+ "\n{\n"
+ "\t\"input_i\" : \"%.2f\",\n"
+ "\t\"input_tp\" : \"%.2f\",\n"
+ "\t\"input_lra\" : \"%.2f\",\n"
+ "\t\"input_thresh\" : \"%.2f\",\n"
+ "\t\"output_i\" : \"%.2f\",\n"
+ "\t\"output_tp\" : \"%+.2f\",\n"
+ "\t\"output_lra\" : \"%.2f\",\n"
+ "\t\"output_thresh\" : \"%.2f\",\n"
+ "\t\"normalization_type\" : \"%s\",\n"
+ "\t\"target_offset\" : \"%.2f\"\n"
+ "}\n",
+ r128_in->integrated_loudness,
+ r128_in->true_peak,
+ r128_in->loudness_range,
+ r128_in->i3000.rel_threshold,
+ r128_out->integrated_loudness,
+ r128_out->true_peak,
+ r128_out->loudness_range,
+ r128_out->i3000.rel_threshold,
+ s->linear_mode ? "linear" : "dynamic",
+ s->target_i - r128_out->integrated_loudness
+ );
+ break;
+
+ case SUMMARY:
+ av_log(ctx, AV_LOG_INFO,
+ "\n"
+ "Input Integrated: %+6.1f LUFS\n"
+ "Input True Peak: %+6.1f dBTP\n"
+ "Input LRA: %6.1f LU\n"
+ "Input Threshold: %+6.1f LUFS\n"
+ "\n"
+ "Output Integrated: %+6.1f LUFS\n"
+ "Output True Peak: %+6.1f dBTP\n"
+ "Output LRA: %6.1f LU\n"
+ "Output Threshold: %+6.1f LUFS\n"
+ "\n"
+ "Normalization Type: %s\n"
+ "Target Offset: %+6.1f LU\n",
+ r128_in->integrated_loudness,
+ r128_in->true_peak,
+ r128_in->loudness_range,
+ r128_in->i3000.rel_threshold,
+ r128_out->integrated_loudness,
+ r128_out->true_peak,
+ r128_out->loudness_range,
+ r128_out->i3000.rel_threshold,
+ s->linear_mode ? "Linear" : "Dynamic",
+ s->target_i - r128_out->integrated_loudness
+ );
+ break;
+ }
+ }
+
+ uninit_ebur128(ctx, &s->r128_in);
+ uninit_ebur128(ctx, &s->r128_out);
+}
+
+static int loudnorm_config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ LoudNormContext *s = ctx->priv;
+ int ret;
+
+ s->nb_samples = FFMAX(inlink->sample_rate / 10, 1);
+
+ ret = config_audio_in(inlink, &s->r128_in);
+ if (ret < 0)
+ return ret;
+
+ ret = config_audio_in(inlink, &s->r128_out);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int loudnorm_config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ LoudNormContext *s = ctx->priv;
+ int ret;
+
+ s->nb_channels = outlink->ch_layout.nb_channels;
+ s->r128_out.peak_mode = PEAK_MODE_TRUE_PEAKS;
+ s->r128_in.peak_mode = PEAK_MODE_TRUE_PEAKS;
+
+ ret = config_audio_out(outlink, &s->r128_in);
+ if (ret < 0)
+ return ret;
+
+ ret = config_audio_out(outlink, &s->r128_out);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int loudnorm_filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ LoudNormContext *s = ctx->priv;
+ EBUR128Context *r128_out = &s->r128_out;
+ EBUR128Context *r128_in = &s->r128_in;
+ const int nb_channels = s->nb_channels;
+ const int nb_samples = in->nb_samples;
+ const double *samples = (const double *)in->data[0];
+ AVFrame *out;
+ int ret;
+
+ ret = true_peaks_ebur128(r128_in, (const uint8_t **)in->data, nb_samples);
+ if (ret < 0)
+ return ret;
+
+ for (int idx_insample = r128_in->idx_insample; idx_insample < nb_samples; idx_insample++) {
+ process_ebur128(r128_in, samples + idx_insample * nb_channels);
+ if (++r128_in->sample_count == inlink->sample_rate / 10) {
+ double loudness_400, loudness_3000;
+ ebur128_loudness(inlink, r128_in, &loudness_400, &loudness_3000);
+ }
+ }
+
+ r128_in->idx_insample = 0;
+ s->insamples = NULL;
+
+ out = ff_get_audio_buffer(outlink, nb_samples);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+
+ if (s->linear_mode) {
+ const double *src = (const double *)in->data[0];
+ double *dst = (double *)out->data[0];
+ const double offset = s->offset;
+
+ for (int n = 0; n < nb_samples * nb_channels; n++)
+ dst[n] = src[n] * offset;
+ } else {
+ }
+
+ ret = true_peaks_ebur128(r128_out, (const uint8_t **)out->data, nb_samples);
+ if (ret < 0)
+ return ret;
+
+ samples = (const double *)out->data[0];
+ for (int idx_insample = r128_out->idx_insample; idx_insample < nb_samples; idx_insample++) {
+ process_ebur128(r128_out, samples + idx_insample * nb_channels);
+ if (++r128_out->sample_count == inlink->sample_rate / 10) {
+ double loudness_400, loudness_3000;
+ ebur128_loudness(inlink, r128_out, &loudness_400, &loudness_3000);
+ }
+ }
+
+ r128_out->idx_insample = 0;
+ av_frame_copy_props(out, in);
+ av_frame_free(&in);
+
+ return ff_filter_frame(outlink, out);
+}
+
+static int loudnorm_activate(AVFilterContext *ctx)
+{
+ AVFilterLink *outlink = ctx->outputs[0];
+ AVFilterLink *inlink = ctx->inputs[0];
+ LoudNormContext *s = ctx->priv;
+ int ret;
+
+ FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
+
+ if (!s->insamples) {
+ AVFrame *in;
+
+ ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in);
+ if (ret < 0)
+ return ret;
+ if (ret > 0)
+ s->insamples = in;
+ }
+
+ if (s->insamples)
+ ret = loudnorm_filter_frame(inlink, s->insamples);
+
+ FF_FILTER_FORWARD_STATUS(inlink, outlink);
+ FF_FILTER_FORWARD_WANTED(outlink, inlink);
+
+ return ret;
+}
+
+#undef OFFSET
+#undef FLAGS
+
+#define OFFSET(x) offsetof(LoudNormContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption loudnorm_options[] = {
+ { "I", "set integrated loudness target", OFFSET(target_i), AV_OPT_TYPE_DOUBLE, {.dbl = -24.}, -70., -5., FLAGS },
+ { "i", "set integrated loudness target", OFFSET(target_i), AV_OPT_TYPE_DOUBLE, {.dbl = -24.}, -70., -5., FLAGS },
+ { "LRA", "set loudness range target", OFFSET(target_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 7.}, 1., 50., FLAGS },
+ { "lra", "set loudness range target", OFFSET(target_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 7.}, 1., 50., FLAGS },
+ { "TP", "set maximum true peak", OFFSET(target_tp), AV_OPT_TYPE_DOUBLE, {.dbl = -2.}, -9., 0., FLAGS },
+ { "tp", "set maximum true peak", OFFSET(target_tp), AV_OPT_TYPE_DOUBLE, {.dbl = -2.}, -9., 0., FLAGS },
+ { "measured_I", "measured IL of input file", OFFSET(measured_i), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 0., FLAGS },
+ { "measured_i", "measured IL of input file", OFFSET(measured_i), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 0., FLAGS },
+ { "measured_LRA", "measured LRA of input file", OFFSET(measured_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, 0., 99., FLAGS },
+ { "measured_lra", "measured LRA of input file", OFFSET(measured_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, 0., 99., FLAGS },
+ { "measured_TP", "measured true peak of input file", OFFSET(measured_tp), AV_OPT_TYPE_DOUBLE, {.dbl = 99.}, -99., 99., FLAGS },
+ { "measured_tp", "measured true peak of input file", OFFSET(measured_tp), AV_OPT_TYPE_DOUBLE, {.dbl = 99.}, -99., 99., FLAGS },
+ { "measured_thresh", "measured threshold of input file", OFFSET(measured_thresh), AV_OPT_TYPE_DOUBLE, {.dbl = -70.}, -99., 0., FLAGS },
+ { "offset", "set offset gain", OFFSET(offset), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 99., FLAGS },
+ { "linear", "normalize linearly if possible", OFFSET(linear_mode), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
+ { "dual_mono", "treat mono input as dual-mono", OFFSET(dual_mono), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
+ { "print_format", "set print format for stats", OFFSET(print_format), AV_OPT_TYPE_INT, {.i64 = NONE}, NONE, PF_NB -1, FLAGS, "print_format" },
+ { "none", 0, 0, AV_OPT_TYPE_CONST, {.i64 = NONE}, 0, 0, FLAGS, "print_format" },
+ { "json", 0, 0, AV_OPT_TYPE_CONST, {.i64 = JSON}, 0, 0, FLAGS, "print_format" },
+ { "summary", 0, 0, AV_OPT_TYPE_CONST, {.i64 = SUMMARY}, 0, 0, FLAGS, "print_format" },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(loudnorm);
+
+static const AVFilterPad loudnorm_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = loudnorm_config_input,
+ },
+};
+
+static const AVFilterPad loudnorm_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = loudnorm_config_output,
+ },
+};
+
+const AVFilter ff_af_loudnorm = {
+ .name = "loudnorm",
+ .description = NULL_IF_CONFIG_SMALL("EBU R128 loudness normalization"),
+ .priv_size = sizeof(LoudNormContext),
+ .priv_class = &loudnorm_class,
+ .init = loudnorm_init,
+ .activate = loudnorm_activate,
+ .uninit = loudnorm_uninit,
+ FILTER_INPUTS(loudnorm_inputs),
+ FILTER_OUTPUTS(loudnorm_outputs),
+ FILTER_QUERY_FUNC(loudnorm_query_formats),
+};
--
2.42.0
[-- Attachment #3: Type: text/plain, Size: 251 bytes --]
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next reply other threads:[~2023-09-29 21:46 UTC|newest]
Thread overview: 22+ messages / expand[flat|nested] mbox.gz Atom feed top
2023-09-29 21:46 Paul B Mahol [this message]
2023-11-15 20:46 ` Paul B Mahol
2023-11-17 6:38 ` Kyle Swanson
2023-11-19 11:56 ` Paul B Mahol
2023-11-19 21:55 ` Marton Balint
2023-11-19 23:37 ` Paul B Mahol
2023-11-21 18:53 ` Kyle Swanson
2023-11-28 16:51 ` Paul B Mahol
2023-11-30 11:43 ` Anton Khirnov
2023-11-30 12:48 ` Paul B Mahol
2023-11-30 13:47 ` Anton Khirnov
2023-11-30 14:01 ` Paul B Mahol
2023-11-30 13:57 ` Anton Khirnov
2023-11-30 14:20 ` Paul B Mahol
2023-11-30 18:34 ` Kyle Swanson
2023-11-30 21:44 ` Paul B Mahol
2023-11-30 22:19 ` Kyle Swanson
2023-11-30 22:51 ` Paul B Mahol
2023-11-30 23:29 ` Kyle Swanson
2023-12-01 10:45 ` Paul B Mahol
2023-12-01 21:12 ` Kyle Swanson
2023-12-01 21:27 ` Paul B Mahol
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