/*
* 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 "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;
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., 20., FLAGS },
{ "lra", "set loudness range target", OFFSET(target_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 7.}, 1., 20., 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(inlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
}
out->pts = s->pts;
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->channels; c++) {
double tmp;
ff_ebur128_sample_peak(s->r128_in, c, &tmp);
if (c == 0 || tmp > true_peak)
true_peak = tmp;
}
offset = s->target_i - global;
offset_tp = true_peak + offset;
s->offset = offset_tp < s->target_tp ? offset : s->target_tp - true_peak;
s->offset = pow(10., s->offset / 20.);
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->channels; c++) {
buf[s->buf_index + c] = src[c];
}
src += inlink->channels;
s->buf_index += inlink->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->channels); n++) {
for (c = 0; c < inlink->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->channels;
if (s->limiter_buf_index >= s->limiter_buf_size)
s->limiter_buf_index -= s->limiter_buf_size;
s->buf_index += inlink->channels;
}
subframe_length = frame_size(inlink->sample_rate, 100);
true_peak_limiter(s, dst, subframe_length, inlink->channels);
ff_ebur128_add_frames_double(s->r128_out, dst, subframe_length);
s->pts +=
out->nb_samples =
inlink->min_samples =
inlink->max_samples =
inlink->partial_buf_size = 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->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->channels;
s->limiter_buf_index += inlink->channels;
if (s->limiter_buf_index >= s->limiter_buf_size)
s->limiter_buf_index -= s->limiter_buf_size;
s->prev_buf_index += inlink->channels;
if (s->prev_buf_index >= s->buf_size)
s->prev_buf_index -= s->buf_size;
s->buf_index += inlink->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->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->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;
s->pts += 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->channels; n++) {
for (c = 0; c < inlink->channels; c++) {
s->limiter_buf[s->limiter_buf_index + c] = src[src_index + c] * gain * s->offset;
}
src_index += inlink->channels;
s->limiter_buf_index += inlink->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->channels);
for (n = 0; n < subframe_length; n++) {
for (c = 0; c < inlink->channels; c++) {
if (src_index < (in->nb_samples * inlink->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->channels))
src_index += inlink->channels;
s->limiter_buf_index += inlink->channels;
if (s->limiter_buf_index >= s->limiter_buf_size)
s->limiter_buf_index -= s->limiter_buf_size;
}
dst += (subframe_length * inlink->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->channels; c++) {
dst[c] = src[c] * s->offset;
}
src += inlink->channels;
dst += inlink->channels;
}
dst = (double *)out->data[0];
ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples);
s->pts += in->nb_samples;
break;
}
if (in != out)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static int request_frame(AVFilterLink *outlink)
{
int ret;
AVFilterContext *ctx = outlink->src;
AVFilterLink *inlink = ctx->inputs[0];
LoudNormContext *s = ctx->priv;
ret = ff_request_frame(inlink);
if (ret == AVERROR_EOF && s->frame_type == INNER_FRAME) {
double *src;
double *buf;
int nb_samples, n, c, offset;
AVFrame *frame;
nb_samples = (s->buf_size / inlink->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->channels) - s->prev_nb_samples) * inlink->channels;
offset -= (frame_size(inlink->sample_rate, 100) - s->prev_nb_samples) * inlink->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->channels; c++) {
src[c] = buf[s->buf_index + c];
}
src += inlink->channels;
s->buf_index += inlink->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 query_formats(AVFilterContext *ctx)
{
LoudNormContext *s = ctx->priv;
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
static const int input_srate[] = {192000, -1};
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_NONE
};
int ret;
layouts = ff_all_channel_counts();
if (!layouts)
return AVERROR(ENOMEM);
ret = ff_set_common_channel_layouts(ctx, layouts);
if (ret < 0)
return ret;
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_set_common_formats(ctx, formats);
if (ret < 0)
return ret;
if (s->frame_type != LINEAR_MODE) {
formats = ff_make_format_list(input_srate);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_formats_ref(formats, &inlink->out_samplerates);
if (ret < 0)
return ret;
ret = ff_formats_ref(formats, &outlink->in_samplerates);
if (ret < 0)
return ret;
}
return 0;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
LoudNormContext *s = ctx->priv;
s->r128_in = ff_ebur128_init(inlink->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->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->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->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->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->channels, sizeof(*s->prev_smp));
if (!s->prev_smp)
return AVERROR(ENOMEM);
init_gaussian_filter(s);
if (s->frame_type != LINEAR_MODE) {
inlink->min_samples =
inlink->max_samples =
inlink->partial_buf_size = frame_size(inlink->sample_rate, 3000);
}
s->pts =
s->buf_index =
s->prev_buf_index =
s->limiter_buf_index = 0;
s->channels = inlink->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,
.filter_frame = filter_frame,
},
{ NULL }
};
static const AVFilterPad avfilter_af_loudnorm_outputs[] = {
{
.name = "default",
.request_frame = request_frame,
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter ff_af_loudnorm = {
.name = "loudnorm",
.description = NULL_IF_CONFIG_SMALL("EBU R128 loudness normalization"),
.priv_size = sizeof(LoudNormContext),
.priv_class = &loudnorm_class,
.query_formats = query_formats,
.init = init,
.uninit = uninit,
.inputs = avfilter_af_loudnorm_inputs,
.outputs = avfilter_af_loudnorm_outputs,
};