mirror of
https://git.videolan.org/git/ffmpeg.git
synced 2024-08-25 19:01:51 +02:00
931 lines
31 KiB
C
931 lines
31 KiB
C
/*
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* Copyright (c) 2016 Kyle Swanson <k@ylo.ph>.
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/* http://k.ylo.ph/2016/04/04/loudnorm.html */
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "internal.h"
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#include "audio.h"
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#include "ebur128.h"
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enum FrameType {
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FIRST_FRAME,
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INNER_FRAME,
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FINAL_FRAME,
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LINEAR_MODE,
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FRAME_NB
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};
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enum LimiterState {
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OUT,
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ATTACK,
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SUSTAIN,
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RELEASE,
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STATE_NB
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};
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enum PrintFormat {
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NONE,
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JSON,
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SUMMARY,
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PF_NB
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};
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typedef struct LoudNormContext {
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const AVClass *class;
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double target_i;
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double target_lra;
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double target_tp;
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double measured_i;
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double measured_lra;
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double measured_tp;
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double measured_thresh;
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double offset;
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int linear;
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int dual_mono;
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enum PrintFormat print_format;
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double *buf;
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int buf_size;
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int buf_index;
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int prev_buf_index;
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double delta[30];
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double weights[21];
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double prev_delta;
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int index;
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double gain_reduction[2];
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double *limiter_buf;
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double *prev_smp;
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int limiter_buf_index;
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int limiter_buf_size;
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enum LimiterState limiter_state;
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int peak_index;
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int env_index;
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int env_cnt;
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int attack_length;
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int release_length;
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int64_t pts;
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enum FrameType frame_type;
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int above_threshold;
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int prev_nb_samples;
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int channels;
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FFEBUR128State *r128_in;
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FFEBUR128State *r128_out;
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} LoudNormContext;
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#define OFFSET(x) offsetof(LoudNormContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption loudnorm_options[] = {
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{ "I", "set integrated loudness target", OFFSET(target_i), AV_OPT_TYPE_DOUBLE, {.dbl = -24.}, -70., -5., FLAGS },
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{ "i", "set integrated loudness target", OFFSET(target_i), AV_OPT_TYPE_DOUBLE, {.dbl = -24.}, -70., -5., FLAGS },
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{ "LRA", "set loudness range target", OFFSET(target_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 7.}, 1., 20., FLAGS },
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{ "lra", "set loudness range target", OFFSET(target_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 7.}, 1., 20., FLAGS },
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{ "TP", "set maximum true peak", OFFSET(target_tp), AV_OPT_TYPE_DOUBLE, {.dbl = -2.}, -9., 0., FLAGS },
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{ "tp", "set maximum true peak", OFFSET(target_tp), AV_OPT_TYPE_DOUBLE, {.dbl = -2.}, -9., 0., FLAGS },
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{ "measured_I", "measured IL of input file", OFFSET(measured_i), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 0., FLAGS },
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{ "measured_i", "measured IL of input file", OFFSET(measured_i), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 0., FLAGS },
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{ "measured_LRA", "measured LRA of input file", OFFSET(measured_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, 0., 99., FLAGS },
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{ "measured_lra", "measured LRA of input file", OFFSET(measured_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, 0., 99., FLAGS },
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{ "measured_TP", "measured true peak of input file", OFFSET(measured_tp), AV_OPT_TYPE_DOUBLE, {.dbl = 99.}, -99., 99., FLAGS },
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{ "measured_tp", "measured true peak of input file", OFFSET(measured_tp), AV_OPT_TYPE_DOUBLE, {.dbl = 99.}, -99., 99., FLAGS },
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{ "measured_thresh", "measured threshold of input file", OFFSET(measured_thresh), AV_OPT_TYPE_DOUBLE, {.dbl = -70.}, -99., 0., FLAGS },
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{ "offset", "set offset gain", OFFSET(offset), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 99., FLAGS },
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{ "linear", "normalize linearly if possible", OFFSET(linear), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
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{ "dual_mono", "treat mono input as dual-mono", OFFSET(dual_mono), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
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{ "print_format", "set print format for stats", OFFSET(print_format), AV_OPT_TYPE_INT, {.i64 = NONE}, NONE, PF_NB -1, FLAGS, "print_format" },
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{ "none", 0, 0, AV_OPT_TYPE_CONST, {.i64 = NONE}, 0, 0, FLAGS, "print_format" },
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{ "json", 0, 0, AV_OPT_TYPE_CONST, {.i64 = JSON}, 0, 0, FLAGS, "print_format" },
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{ "summary", 0, 0, AV_OPT_TYPE_CONST, {.i64 = SUMMARY}, 0, 0, FLAGS, "print_format" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(loudnorm);
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static inline int frame_size(int sample_rate, int frame_len_msec)
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{
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const int frame_size = round((double)sample_rate * (frame_len_msec / 1000.0));
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return frame_size + (frame_size % 2);
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}
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static void init_gaussian_filter(LoudNormContext *s)
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{
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double total_weight = 0.0;
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const double sigma = 3.5;
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double adjust;
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int i;
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const int offset = 21 / 2;
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const double c1 = 1.0 / (sigma * sqrt(2.0 * M_PI));
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const double c2 = 2.0 * pow(sigma, 2.0);
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for (i = 0; i < 21; i++) {
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const int x = i - offset;
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s->weights[i] = c1 * exp(-(pow(x, 2.0) / c2));
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total_weight += s->weights[i];
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}
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adjust = 1.0 / total_weight;
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for (i = 0; i < 21; i++)
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s->weights[i] *= adjust;
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}
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static double gaussian_filter(LoudNormContext *s, int index)
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{
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double result = 0.;
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int i;
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index = index - 10 > 0 ? index - 10 : index + 20;
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for (i = 0; i < 21; i++)
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result += s->delta[((index + i) < 30) ? (index + i) : (index + i - 30)] * s->weights[i];
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return result;
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}
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static void detect_peak(LoudNormContext *s, int offset, int nb_samples, int channels, int *peak_delta, double *peak_value)
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{
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int n, c, i, index;
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double ceiling;
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double *buf;
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*peak_delta = -1;
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buf = s->limiter_buf;
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ceiling = s->target_tp;
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index = s->limiter_buf_index + (offset * channels) + (1920 * channels);
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if (index >= s->limiter_buf_size)
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index -= s->limiter_buf_size;
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if (s->frame_type == FIRST_FRAME) {
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for (c = 0; c < channels; c++)
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s->prev_smp[c] = fabs(buf[index + c - channels]);
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}
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for (n = 0; n < nb_samples; n++) {
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for (c = 0; c < channels; c++) {
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double this, next, max_peak;
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this = fabs(buf[(index + c) < s->limiter_buf_size ? (index + c) : (index + c - s->limiter_buf_size)]);
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next = fabs(buf[(index + c + channels) < s->limiter_buf_size ? (index + c + channels) : (index + c + channels - s->limiter_buf_size)]);
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if ((s->prev_smp[c] <= this) && (next <= this) && (this > ceiling) && (n > 0)) {
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int detected;
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detected = 1;
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for (i = 2; i < 12; i++) {
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next = fabs(buf[(index + c + (i * channels)) < s->limiter_buf_size ? (index + c + (i * channels)) : (index + c + (i * channels) - s->limiter_buf_size)]);
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if (next > this) {
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detected = 0;
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break;
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}
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}
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if (!detected)
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continue;
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for (c = 0; c < channels; c++) {
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if (c == 0 || fabs(buf[index + c]) > max_peak)
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max_peak = fabs(buf[index + c]);
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s->prev_smp[c] = fabs(buf[(index + c) < s->limiter_buf_size ? (index + c) : (index + c - s->limiter_buf_size)]);
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}
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*peak_delta = n;
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s->peak_index = index;
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*peak_value = max_peak;
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return;
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}
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s->prev_smp[c] = this;
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}
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index += channels;
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if (index >= s->limiter_buf_size)
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index -= s->limiter_buf_size;
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}
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}
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static void true_peak_limiter(LoudNormContext *s, double *out, int nb_samples, int channels)
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{
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int n, c, index, peak_delta, smp_cnt;
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double ceiling, peak_value;
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double *buf;
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buf = s->limiter_buf;
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ceiling = s->target_tp;
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index = s->limiter_buf_index;
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smp_cnt = 0;
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if (s->frame_type == FIRST_FRAME) {
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double max;
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max = 0.;
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for (n = 0; n < 1920; n++) {
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for (c = 0; c < channels; c++) {
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max = fabs(buf[c]) > max ? fabs(buf[c]) : max;
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}
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buf += channels;
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}
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if (max > ceiling) {
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s->gain_reduction[1] = ceiling / max;
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s->limiter_state = SUSTAIN;
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buf = s->limiter_buf;
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for (n = 0; n < 1920; n++) {
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for (c = 0; c < channels; c++) {
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double env;
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env = s->gain_reduction[1];
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buf[c] *= env;
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}
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buf += channels;
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}
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}
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buf = s->limiter_buf;
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}
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do {
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switch(s->limiter_state) {
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case OUT:
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detect_peak(s, smp_cnt, nb_samples - smp_cnt, channels, &peak_delta, &peak_value);
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if (peak_delta != -1) {
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s->env_cnt = 0;
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smp_cnt += (peak_delta - s->attack_length);
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s->gain_reduction[0] = 1.;
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s->gain_reduction[1] = ceiling / peak_value;
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s->limiter_state = ATTACK;
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s->env_index = s->peak_index - (s->attack_length * channels);
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if (s->env_index < 0)
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s->env_index += s->limiter_buf_size;
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s->env_index += (s->env_cnt * channels);
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if (s->env_index > s->limiter_buf_size)
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s->env_index -= s->limiter_buf_size;
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} else {
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smp_cnt = nb_samples;
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}
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break;
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case ATTACK:
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for (; s->env_cnt < s->attack_length; s->env_cnt++) {
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for (c = 0; c < channels; c++) {
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double env;
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env = s->gain_reduction[0] - ((double) s->env_cnt / (s->attack_length - 1) * (s->gain_reduction[0] - s->gain_reduction[1]));
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buf[s->env_index + c] *= env;
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}
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s->env_index += channels;
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if (s->env_index >= s->limiter_buf_size)
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s->env_index -= s->limiter_buf_size;
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smp_cnt++;
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if (smp_cnt >= nb_samples) {
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s->env_cnt++;
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break;
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}
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}
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if (smp_cnt < nb_samples) {
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s->env_cnt = 0;
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s->attack_length = 1920;
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s->limiter_state = SUSTAIN;
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}
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break;
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case SUSTAIN:
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detect_peak(s, smp_cnt, nb_samples, channels, &peak_delta, &peak_value);
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if (peak_delta == -1) {
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s->limiter_state = RELEASE;
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s->gain_reduction[0] = s->gain_reduction[1];
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s->gain_reduction[1] = 1.;
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s->env_cnt = 0;
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break;
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} else {
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double gain_reduction;
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gain_reduction = ceiling / peak_value;
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if (gain_reduction < s->gain_reduction[1]) {
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s->limiter_state = ATTACK;
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s->attack_length = peak_delta;
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if (s->attack_length <= 1)
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s->attack_length = 2;
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s->gain_reduction[0] = s->gain_reduction[1];
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s->gain_reduction[1] = gain_reduction;
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s->env_cnt = 0;
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break;
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}
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for (s->env_cnt = 0; s->env_cnt < peak_delta; s->env_cnt++) {
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for (c = 0; c < channels; c++) {
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double env;
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env = s->gain_reduction[1];
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buf[s->env_index + c] *= env;
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}
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s->env_index += channels;
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if (s->env_index >= s->limiter_buf_size)
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s->env_index -= s->limiter_buf_size;
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smp_cnt++;
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if (smp_cnt >= nb_samples) {
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s->env_cnt++;
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break;
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}
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}
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}
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break;
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case RELEASE:
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for (; s->env_cnt < s->release_length; s->env_cnt++) {
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for (c = 0; c < channels; c++) {
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double env;
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env = s->gain_reduction[0] + (((double) s->env_cnt / (s->release_length - 1)) * (s->gain_reduction[1] - s->gain_reduction[0]));
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buf[s->env_index + c] *= env;
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}
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s->env_index += channels;
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if (s->env_index >= s->limiter_buf_size)
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s->env_index -= s->limiter_buf_size;
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smp_cnt++;
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if (smp_cnt >= nb_samples) {
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s->env_cnt++;
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break;
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}
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}
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if (smp_cnt < nb_samples) {
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s->env_cnt = 0;
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s->limiter_state = OUT;
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}
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break;
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}
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} while (smp_cnt < nb_samples);
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for (n = 0; n < nb_samples; n++) {
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for (c = 0; c < channels; c++) {
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out[c] = buf[index + c];
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if (fabs(out[c]) > ceiling) {
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out[c] = ceiling * (out[c] < 0 ? -1 : 1);
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}
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}
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out += channels;
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index += channels;
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if (index >= s->limiter_buf_size)
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index -= s->limiter_buf_size;
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}
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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LoudNormContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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AVFrame *out;
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const double *src;
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double *dst;
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double *buf;
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double *limiter_buf;
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int i, n, c, subframe_length, src_index;
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double gain, gain_next, env_global, env_shortterm,
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global, shortterm, lra, relative_threshold;
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if (av_frame_is_writable(in)) {
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out = in;
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} else {
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out = ff_get_audio_buffer(inlink, in->nb_samples);
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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}
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out->pts = s->pts;
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src = (const double *)in->data[0];
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dst = (double *)out->data[0];
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buf = s->buf;
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limiter_buf = s->limiter_buf;
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ff_ebur128_add_frames_double(s->r128_in, src, in->nb_samples);
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if (s->frame_type == FIRST_FRAME && in->nb_samples < frame_size(inlink->sample_rate, 3000)) {
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double offset, offset_tp, true_peak;
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ff_ebur128_loudness_global(s->r128_in, &global);
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for (c = 0; c < inlink->channels; c++) {
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double tmp;
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ff_ebur128_sample_peak(s->r128_in, c, &tmp);
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if (c == 0 || tmp > true_peak)
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true_peak = tmp;
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}
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offset = s->target_i - global;
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offset_tp = true_peak + offset;
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s->offset = offset_tp < s->target_tp ? offset : s->target_tp - true_peak;
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s->offset = pow(10., s->offset / 20.);
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s->frame_type = LINEAR_MODE;
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}
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switch (s->frame_type) {
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case FIRST_FRAME:
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for (n = 0; n < in->nb_samples; n++) {
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for (c = 0; c < inlink->channels; c++) {
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buf[s->buf_index + c] = src[c];
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}
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src += inlink->channels;
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s->buf_index += inlink->channels;
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}
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ff_ebur128_loudness_shortterm(s->r128_in, &shortterm);
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if (shortterm < s->measured_thresh) {
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s->above_threshold = 0;
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env_shortterm = shortterm <= -70. ? 0. : s->target_i - s->measured_i;
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} else {
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s->above_threshold = 1;
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|
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,
|
|
};
|