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/* |
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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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314 |
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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.; |
327 |
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s->env_cnt = 0; |
328 |
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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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333 |
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if (gain_reduction < s->gain_reduction[1]) { |
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s->limiter_state = ATTACK; |
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336 |
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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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340 |
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s->gain_reduction[0] = s->gain_reduction[1]; |
341 |
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s->gain_reduction[1] = gain_reduction; |
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s->env_cnt = 0; |
343 |
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break; |
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} |
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346 |
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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++) { |
348 |
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double env; |
349 |
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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; |
356 |
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357 |
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smp_cnt++; |
358 |
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if (smp_cnt >= nb_samples) { |
359 |
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s->env_cnt++; |
360 |
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break; |
361 |
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} |
362 |
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} |
363 |
|
|
} |
364 |
|
|
break; |
365 |
|
|
|
366 |
|
|
case RELEASE: |
367 |
|
|
for (; s->env_cnt < s->release_length; s->env_cnt++) { |
368 |
|
|
for (c = 0; c < channels; c++) { |
369 |
|
|
double env; |
370 |
|
|
env = s->gain_reduction[0] + (((double) s->env_cnt / (s->release_length - 1)) * (s->gain_reduction[1] - s->gain_reduction[0])); |
371 |
|
|
buf[s->env_index + c] *= env; |
372 |
|
|
} |
373 |
|
|
|
374 |
|
|
s->env_index += channels; |
375 |
|
|
if (s->env_index >= s->limiter_buf_size) |
376 |
|
|
s->env_index -= s->limiter_buf_size; |
377 |
|
|
|
378 |
|
|
smp_cnt++; |
379 |
|
|
if (smp_cnt >= nb_samples) { |
380 |
|
|
s->env_cnt++; |
381 |
|
|
break; |
382 |
|
|
} |
383 |
|
|
} |
384 |
|
|
|
385 |
|
|
if (smp_cnt < nb_samples) { |
386 |
|
|
s->env_cnt = 0; |
387 |
|
|
s->limiter_state = OUT; |
388 |
|
|
} |
389 |
|
|
|
390 |
|
|
break; |
391 |
|
|
} |
392 |
|
|
|
393 |
|
|
} while (smp_cnt < nb_samples); |
394 |
|
|
|
395 |
|
|
for (n = 0; n < nb_samples; n++) { |
396 |
|
|
for (c = 0; c < channels; c++) { |
397 |
|
|
out[c] = buf[index + c]; |
398 |
|
|
if (fabs(out[c]) > ceiling) { |
399 |
|
|
out[c] = ceiling * (out[c] < 0 ? -1 : 1); |
400 |
|
|
} |
401 |
|
|
} |
402 |
|
|
out += channels; |
403 |
|
|
index += channels; |
404 |
|
|
if (index >= s->limiter_buf_size) |
405 |
|
|
index -= s->limiter_buf_size; |
406 |
|
|
} |
407 |
|
|
} |
408 |
|
|
|
409 |
|
|
static int filter_frame(AVFilterLink *inlink, AVFrame *in) |
410 |
|
|
{ |
411 |
|
|
AVFilterContext *ctx = inlink->dst; |
412 |
|
|
LoudNormContext *s = ctx->priv; |
413 |
|
|
AVFilterLink *outlink = ctx->outputs[0]; |
414 |
|
|
AVFrame *out; |
415 |
|
|
const double *src; |
416 |
|
|
double *dst; |
417 |
|
|
double *buf; |
418 |
|
|
double *limiter_buf; |
419 |
|
|
int i, n, c, subframe_length, src_index; |
420 |
|
|
double gain, gain_next, env_global, env_shortterm, |
421 |
|
|
global, shortterm, lra, relative_threshold; |
422 |
|
|
|
423 |
|
|
if (av_frame_is_writable(in)) { |
424 |
|
|
out = in; |
425 |
|
|
} else { |
426 |
|
|
out = ff_get_audio_buffer(outlink, in->nb_samples); |
427 |
|
|
if (!out) { |
428 |
|
|
av_frame_free(&in); |
429 |
|
|
return AVERROR(ENOMEM); |
430 |
|
|
} |
431 |
|
|
av_frame_copy_props(out, in); |
432 |
|
|
} |
433 |
|
|
|
434 |
|
|
if (s->pts == AV_NOPTS_VALUE) |
435 |
|
|
s->pts = in->pts; |
436 |
|
|
|
437 |
|
|
out->pts = s->pts; |
438 |
|
|
src = (const double *)in->data[0]; |
439 |
|
|
dst = (double *)out->data[0]; |
440 |
|
|
buf = s->buf; |
441 |
|
|
limiter_buf = s->limiter_buf; |
442 |
|
|
|
443 |
|
|
ff_ebur128_add_frames_double(s->r128_in, src, in->nb_samples); |
444 |
|
|
|
445 |
|
|
if (s->frame_type == FIRST_FRAME && in->nb_samples < frame_size(inlink->sample_rate, 3000)) { |
446 |
|
|
double offset, offset_tp, true_peak; |
447 |
|
|
|
448 |
|
|
ff_ebur128_loudness_global(s->r128_in, &global); |
449 |
|
|
for (c = 0; c < inlink->channels; c++) { |
450 |
|
|
double tmp; |
451 |
|
|
ff_ebur128_sample_peak(s->r128_in, c, &tmp); |
452 |
|
|
if (c == 0 || tmp > true_peak) |
453 |
|
|
true_peak = tmp; |
454 |
|
|
} |
455 |
|
|
|
456 |
|
|
offset = pow(10., (s->target_i - global) / 20.); |
457 |
|
|
offset_tp = true_peak * offset; |
458 |
|
|
s->offset = offset_tp < s->target_tp ? offset : s->target_tp - true_peak; |
459 |
|
|
s->frame_type = LINEAR_MODE; |
460 |
|
|
} |
461 |
|
|
|
462 |
|
|
switch (s->frame_type) { |
463 |
|
|
case FIRST_FRAME: |
464 |
|
|
for (n = 0; n < in->nb_samples; n++) { |
465 |
|
|
for (c = 0; c < inlink->channels; c++) { |
466 |
|
|
buf[s->buf_index + c] = src[c]; |
467 |
|
|
} |
468 |
|
|
src += inlink->channels; |
469 |
|
|
s->buf_index += inlink->channels; |
470 |
|
|
} |
471 |
|
|
|
472 |
|
|
ff_ebur128_loudness_shortterm(s->r128_in, &shortterm); |
473 |
|
|
|
474 |
|
|
if (shortterm < s->measured_thresh) { |
475 |
|
|
s->above_threshold = 0; |
476 |
|
|
env_shortterm = shortterm <= -70. ? 0. : s->target_i - s->measured_i; |
477 |
|
|
} else { |
478 |
|
|
s->above_threshold = 1; |
479 |
|
|
env_shortterm = shortterm <= -70. ? 0. : s->target_i - shortterm; |
480 |
|
|
} |
481 |
|
|
|
482 |
|
|
for (n = 0; n < 30; n++) |
483 |
|
|
s->delta[n] = pow(10., env_shortterm / 20.); |
484 |
|
|
s->prev_delta = s->delta[s->index]; |
485 |
|
|
|
486 |
|
|
s->buf_index = |
487 |
|
|
s->limiter_buf_index = 0; |
488 |
|
|
|
489 |
|
|
for (n = 0; n < (s->limiter_buf_size / inlink->channels); n++) { |
490 |
|
|
for (c = 0; c < inlink->channels; c++) { |
491 |
|
|
limiter_buf[s->limiter_buf_index + c] = buf[s->buf_index + c] * s->delta[s->index] * s->offset; |
492 |
|
|
} |
493 |
|
|
s->limiter_buf_index += inlink->channels; |
494 |
|
|
if (s->limiter_buf_index >= s->limiter_buf_size) |
495 |
|
|
s->limiter_buf_index -= s->limiter_buf_size; |
496 |
|
|
|
497 |
|
|
s->buf_index += inlink->channels; |
498 |
|
|
} |
499 |
|
|
|
500 |
|
|
subframe_length = frame_size(inlink->sample_rate, 100); |
501 |
|
|
true_peak_limiter(s, dst, subframe_length, inlink->channels); |
502 |
|
|
ff_ebur128_add_frames_double(s->r128_out, dst, subframe_length); |
503 |
|
|
|
504 |
|
|
s->pts += |
505 |
|
|
out->nb_samples = |
506 |
|
|
inlink->min_samples = |
507 |
|
|
inlink->max_samples = |
508 |
|
|
inlink->partial_buf_size = subframe_length; |
509 |
|
|
|
510 |
|
|
s->frame_type = INNER_FRAME; |
511 |
|
|
break; |
512 |
|
|
|
513 |
|
|
case INNER_FRAME: |
514 |
|
|
gain = gaussian_filter(s, s->index + 10 < 30 ? s->index + 10 : s->index + 10 - 30); |
515 |
|
|
gain_next = gaussian_filter(s, s->index + 11 < 30 ? s->index + 11 : s->index + 11 - 30); |
516 |
|
|
|
517 |
|
|
for (n = 0; n < in->nb_samples; n++) { |
518 |
|
|
for (c = 0; c < inlink->channels; c++) { |
519 |
|
|
buf[s->prev_buf_index + c] = src[c]; |
520 |
|
|
limiter_buf[s->limiter_buf_index + c] = buf[s->buf_index + c] * (gain + (((double) n / in->nb_samples) * (gain_next - gain))) * s->offset; |
521 |
|
|
} |
522 |
|
|
src += inlink->channels; |
523 |
|
|
|
524 |
|
|
s->limiter_buf_index += inlink->channels; |
525 |
|
|
if (s->limiter_buf_index >= s->limiter_buf_size) |
526 |
|
|
s->limiter_buf_index -= s->limiter_buf_size; |
527 |
|
|
|
528 |
|
|
s->prev_buf_index += inlink->channels; |
529 |
|
|
if (s->prev_buf_index >= s->buf_size) |
530 |
|
|
s->prev_buf_index -= s->buf_size; |
531 |
|
|
|
532 |
|
|
s->buf_index += inlink->channels; |
533 |
|
|
if (s->buf_index >= s->buf_size) |
534 |
|
|
s->buf_index -= s->buf_size; |
535 |
|
|
} |
536 |
|
|
|
537 |
|
|
subframe_length = (frame_size(inlink->sample_rate, 100) - in->nb_samples) * inlink->channels; |
538 |
|
|
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; |
539 |
|
|
|
540 |
|
|
true_peak_limiter(s, dst, in->nb_samples, inlink->channels); |
541 |
|
|
ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples); |
542 |
|
|
|
543 |
|
|
ff_ebur128_loudness_range(s->r128_in, &lra); |
544 |
|
|
ff_ebur128_loudness_global(s->r128_in, &global); |
545 |
|
|
ff_ebur128_loudness_shortterm(s->r128_in, &shortterm); |
546 |
|
|
ff_ebur128_relative_threshold(s->r128_in, &relative_threshold); |
547 |
|
|
|
548 |
|
|
if (s->above_threshold == 0) { |
549 |
|
|
double shortterm_out; |
550 |
|
|
|
551 |
|
|
if (shortterm > s->measured_thresh) |
552 |
|
|
s->prev_delta *= 1.0058; |
553 |
|
|
|
554 |
|
|
ff_ebur128_loudness_shortterm(s->r128_out, &shortterm_out); |
555 |
|
|
if (shortterm_out >= s->target_i) |
556 |
|
|
s->above_threshold = 1; |
557 |
|
|
} |
558 |
|
|
|
559 |
|
|
if (shortterm < relative_threshold || shortterm <= -70. || s->above_threshold == 0) { |
560 |
|
|
s->delta[s->index] = s->prev_delta; |
561 |
|
|
} else { |
562 |
|
|
env_global = fabs(shortterm - global) < (s->target_lra / 2.) ? shortterm - global : (s->target_lra / 2.) * ((shortterm - global) < 0 ? -1 : 1); |
563 |
|
|
env_shortterm = s->target_i - shortterm; |
564 |
|
|
s->delta[s->index] = pow(10., (env_global + env_shortterm) / 20.); |
565 |
|
|
} |
566 |
|
|
|
567 |
|
|
s->prev_delta = s->delta[s->index]; |
568 |
|
|
s->index++; |
569 |
|
|
if (s->index >= 30) |
570 |
|
|
s->index -= 30; |
571 |
|
|
s->prev_nb_samples = in->nb_samples; |
572 |
|
|
s->pts += in->nb_samples; |
573 |
|
|
break; |
574 |
|
|
|
575 |
|
|
case FINAL_FRAME: |
576 |
|
|
gain = gaussian_filter(s, s->index + 10 < 30 ? s->index + 10 : s->index + 10 - 30); |
577 |
|
|
s->limiter_buf_index = 0; |
578 |
|
|
src_index = 0; |
579 |
|
|
|
580 |
|
|
for (n = 0; n < s->limiter_buf_size / inlink->channels; n++) { |
581 |
|
|
for (c = 0; c < inlink->channels; c++) { |
582 |
|
|
s->limiter_buf[s->limiter_buf_index + c] = src[src_index + c] * gain * s->offset; |
583 |
|
|
} |
584 |
|
|
src_index += inlink->channels; |
585 |
|
|
|
586 |
|
|
s->limiter_buf_index += inlink->channels; |
587 |
|
|
if (s->limiter_buf_index >= s->limiter_buf_size) |
588 |
|
|
s->limiter_buf_index -= s->limiter_buf_size; |
589 |
|
|
} |
590 |
|
|
|
591 |
|
|
subframe_length = frame_size(inlink->sample_rate, 100); |
592 |
|
|
for (i = 0; i < in->nb_samples / subframe_length; i++) { |
593 |
|
|
true_peak_limiter(s, dst, subframe_length, inlink->channels); |
594 |
|
|
|
595 |
|
|
for (n = 0; n < subframe_length; n++) { |
596 |
|
|
for (c = 0; c < inlink->channels; c++) { |
597 |
|
|
if (src_index < (in->nb_samples * inlink->channels)) { |
598 |
|
|
limiter_buf[s->limiter_buf_index + c] = src[src_index + c] * gain * s->offset; |
599 |
|
|
} else { |
600 |
|
|
limiter_buf[s->limiter_buf_index + c] = 0.; |
601 |
|
|
} |
602 |
|
|
} |
603 |
|
|
|
604 |
|
|
if (src_index < (in->nb_samples * inlink->channels)) |
605 |
|
|
src_index += inlink->channels; |
606 |
|
|
|
607 |
|
|
s->limiter_buf_index += inlink->channels; |
608 |
|
|
if (s->limiter_buf_index >= s->limiter_buf_size) |
609 |
|
|
s->limiter_buf_index -= s->limiter_buf_size; |
610 |
|
|
} |
611 |
|
|
|
612 |
|
|
dst += (subframe_length * inlink->channels); |
613 |
|
|
} |
614 |
|
|
|
615 |
|
|
dst = (double *)out->data[0]; |
616 |
|
|
ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples); |
617 |
|
|
break; |
618 |
|
|
|
619 |
|
|
case LINEAR_MODE: |
620 |
|
|
for (n = 0; n < in->nb_samples; n++) { |
621 |
|
|
for (c = 0; c < inlink->channels; c++) { |
622 |
|
|
dst[c] = src[c] * s->offset; |
623 |
|
|
} |
624 |
|
|
src += inlink->channels; |
625 |
|
|
dst += inlink->channels; |
626 |
|
|
} |
627 |
|
|
|
628 |
|
|
dst = (double *)out->data[0]; |
629 |
|
|
ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples); |
630 |
|
|
s->pts += in->nb_samples; |
631 |
|
|
break; |
632 |
|
|
} |
633 |
|
|
|
634 |
|
|
if (in != out) |
635 |
|
|
av_frame_free(&in); |
636 |
|
|
|
637 |
|
|
return ff_filter_frame(outlink, out); |
638 |
|
|
} |
639 |
|
|
|
640 |
|
|
static int request_frame(AVFilterLink *outlink) |
641 |
|
|
{ |
642 |
|
|
int ret; |
643 |
|
|
AVFilterContext *ctx = outlink->src; |
644 |
|
|
AVFilterLink *inlink = ctx->inputs[0]; |
645 |
|
|
LoudNormContext *s = ctx->priv; |
646 |
|
|
|
647 |
|
|
ret = ff_request_frame(inlink); |
648 |
|
|
if (ret == AVERROR_EOF && s->frame_type == INNER_FRAME) { |
649 |
|
|
double *src; |
650 |
|
|
double *buf; |
651 |
|
|
int nb_samples, n, c, offset; |
652 |
|
|
AVFrame *frame; |
653 |
|
|
|
654 |
|
|
nb_samples = (s->buf_size / inlink->channels) - s->prev_nb_samples; |
655 |
|
|
nb_samples -= (frame_size(inlink->sample_rate, 100) - s->prev_nb_samples); |
656 |
|
|
|
657 |
|
|
frame = ff_get_audio_buffer(outlink, nb_samples); |
658 |
|
|
if (!frame) |
659 |
|
|
return AVERROR(ENOMEM); |
660 |
|
|
frame->nb_samples = nb_samples; |
661 |
|
|
|
662 |
|
|
buf = s->buf; |
663 |
|
|
src = (double *)frame->data[0]; |
664 |
|
|
|
665 |
|
|
offset = ((s->limiter_buf_size / inlink->channels) - s->prev_nb_samples) * inlink->channels; |
666 |
|
|
offset -= (frame_size(inlink->sample_rate, 100) - s->prev_nb_samples) * inlink->channels; |
667 |
|
|
s->buf_index = s->buf_index - offset < 0 ? s->buf_index - offset + s->buf_size : s->buf_index - offset; |
668 |
|
|
|
669 |
|
|
for (n = 0; n < nb_samples; n++) { |
670 |
|
|
for (c = 0; c < inlink->channels; c++) { |
671 |
|
|
src[c] = buf[s->buf_index + c]; |
672 |
|
|
} |
673 |
|
|
src += inlink->channels; |
674 |
|
|
s->buf_index += inlink->channels; |
675 |
|
|
if (s->buf_index >= s->buf_size) |
676 |
|
|
s->buf_index -= s->buf_size; |
677 |
|
|
} |
678 |
|
|
|
679 |
|
|
s->frame_type = FINAL_FRAME; |
680 |
|
|
ret = filter_frame(inlink, frame); |
681 |
|
|
} |
682 |
|
|
return ret; |
683 |
|
|
} |
684 |
|
|
|
685 |
|
|
static int query_formats(AVFilterContext *ctx) |
686 |
|
|
{ |
687 |
|
|
LoudNormContext *s = ctx->priv; |
688 |
|
|
AVFilterFormats *formats; |
689 |
|
|
AVFilterChannelLayouts *layouts; |
690 |
|
|
AVFilterLink *inlink = ctx->inputs[0]; |
691 |
|
|
AVFilterLink *outlink = ctx->outputs[0]; |
692 |
|
|
static const int input_srate[] = {192000, -1}; |
693 |
|
|
static const enum AVSampleFormat sample_fmts[] = { |
694 |
|
|
AV_SAMPLE_FMT_DBL, |
695 |
|
|
AV_SAMPLE_FMT_NONE |
696 |
|
|
}; |
697 |
|
|
int ret; |
698 |
|
|
|
699 |
|
|
layouts = ff_all_channel_counts(); |
700 |
|
|
if (!layouts) |
701 |
|
|
return AVERROR(ENOMEM); |
702 |
|
|
ret = ff_set_common_channel_layouts(ctx, layouts); |
703 |
|
|
if (ret < 0) |
704 |
|
|
return ret; |
705 |
|
|
|
706 |
|
|
formats = ff_make_format_list(sample_fmts); |
707 |
|
|
if (!formats) |
708 |
|
|
return AVERROR(ENOMEM); |
709 |
|
|
ret = ff_set_common_formats(ctx, formats); |
710 |
|
|
if (ret < 0) |
711 |
|
|
return ret; |
712 |
|
|
|
713 |
|
|
if (s->frame_type != LINEAR_MODE) { |
714 |
|
|
formats = ff_make_format_list(input_srate); |
715 |
|
|
if (!formats) |
716 |
|
|
return AVERROR(ENOMEM); |
717 |
|
|
ret = ff_formats_ref(formats, &inlink->outcfg.samplerates); |
718 |
|
|
if (ret < 0) |
719 |
|
|
return ret; |
720 |
|
|
ret = ff_formats_ref(formats, &outlink->incfg.samplerates); |
721 |
|
|
if (ret < 0) |
722 |
|
|
return ret; |
723 |
|
|
} |
724 |
|
|
|
725 |
|
|
return 0; |
726 |
|
|
} |
727 |
|
|
|
728 |
|
|
static int config_input(AVFilterLink *inlink) |
729 |
|
|
{ |
730 |
|
|
AVFilterContext *ctx = inlink->dst; |
731 |
|
|
LoudNormContext *s = ctx->priv; |
732 |
|
|
|
733 |
|
|
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); |
734 |
|
|
if (!s->r128_in) |
735 |
|
|
return AVERROR(ENOMEM); |
736 |
|
|
|
737 |
|
|
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); |
738 |
|
|
if (!s->r128_out) |
739 |
|
|
return AVERROR(ENOMEM); |
740 |
|
|
|
741 |
|
|
if (inlink->channels == 1 && s->dual_mono) { |
742 |
|
|
ff_ebur128_set_channel(s->r128_in, 0, FF_EBUR128_DUAL_MONO); |
743 |
|
|
ff_ebur128_set_channel(s->r128_out, 0, FF_EBUR128_DUAL_MONO); |
744 |
|
|
} |
745 |
|
|
|
746 |
|
|
s->buf_size = frame_size(inlink->sample_rate, 3000) * inlink->channels; |
747 |
|
|
s->buf = av_malloc_array(s->buf_size, sizeof(*s->buf)); |
748 |
|
|
if (!s->buf) |
749 |
|
|
return AVERROR(ENOMEM); |
750 |
|
|
|
751 |
|
|
s->limiter_buf_size = frame_size(inlink->sample_rate, 210) * inlink->channels; |
752 |
|
|
s->limiter_buf = av_malloc_array(s->buf_size, sizeof(*s->limiter_buf)); |
753 |
|
|
if (!s->limiter_buf) |
754 |
|
|
return AVERROR(ENOMEM); |
755 |
|
|
|
756 |
|
|
s->prev_smp = av_malloc_array(inlink->channels, sizeof(*s->prev_smp)); |
757 |
|
|
if (!s->prev_smp) |
758 |
|
|
return AVERROR(ENOMEM); |
759 |
|
|
|
760 |
|
|
init_gaussian_filter(s); |
761 |
|
|
|
762 |
|
|
if (s->frame_type != LINEAR_MODE) { |
763 |
|
|
inlink->min_samples = |
764 |
|
|
inlink->max_samples = |
765 |
|
|
inlink->partial_buf_size = frame_size(inlink->sample_rate, 3000); |
766 |
|
|
} |
767 |
|
|
|
768 |
|
|
s->pts = AV_NOPTS_VALUE; |
769 |
|
|
s->buf_index = |
770 |
|
|
s->prev_buf_index = |
771 |
|
|
s->limiter_buf_index = 0; |
772 |
|
|
s->channels = inlink->channels; |
773 |
|
|
s->index = 1; |
774 |
|
|
s->limiter_state = OUT; |
775 |
|
|
s->offset = pow(10., s->offset / 20.); |
776 |
|
|
s->target_tp = pow(10., s->target_tp / 20.); |
777 |
|
|
s->attack_length = frame_size(inlink->sample_rate, 10); |
778 |
|
|
s->release_length = frame_size(inlink->sample_rate, 100); |
779 |
|
|
|
780 |
|
|
return 0; |
781 |
|
|
} |
782 |
|
|
|
783 |
|
|
static av_cold int init(AVFilterContext *ctx) |
784 |
|
|
{ |
785 |
|
|
LoudNormContext *s = ctx->priv; |
786 |
|
|
s->frame_type = FIRST_FRAME; |
787 |
|
|
|
788 |
|
|
if (s->linear) { |
789 |
|
|
double offset, offset_tp; |
790 |
|
|
offset = s->target_i - s->measured_i; |
791 |
|
|
offset_tp = s->measured_tp + offset; |
792 |
|
|
|
793 |
|
|
if (s->measured_tp != 99 && s->measured_thresh != -70 && s->measured_lra != 0 && s->measured_i != 0) { |
794 |
|
|
if ((offset_tp <= s->target_tp) && (s->measured_lra <= s->target_lra)) { |
795 |
|
|
s->frame_type = LINEAR_MODE; |
796 |
|
|
s->offset = offset; |
797 |
|
|
} |
798 |
|
|
} |
799 |
|
|
} |
800 |
|
|
|
801 |
|
|
return 0; |
802 |
|
|
} |
803 |
|
|
|
804 |
|
|
static av_cold void uninit(AVFilterContext *ctx) |
805 |
|
|
{ |
806 |
|
|
LoudNormContext *s = ctx->priv; |
807 |
|
|
double i_in, i_out, lra_in, lra_out, thresh_in, thresh_out, tp_in, tp_out; |
808 |
|
|
int c; |
809 |
|
|
|
810 |
|
|
if (!s->r128_in || !s->r128_out) |
811 |
|
|
goto end; |
812 |
|
|
|
813 |
|
|
ff_ebur128_loudness_range(s->r128_in, &lra_in); |
814 |
|
|
ff_ebur128_loudness_global(s->r128_in, &i_in); |
815 |
|
|
ff_ebur128_relative_threshold(s->r128_in, &thresh_in); |
816 |
|
|
for (c = 0; c < s->channels; c++) { |
817 |
|
|
double tmp; |
818 |
|
|
ff_ebur128_sample_peak(s->r128_in, c, &tmp); |
819 |
|
|
if ((c == 0) || (tmp > tp_in)) |
820 |
|
|
tp_in = tmp; |
821 |
|
|
} |
822 |
|
|
|
823 |
|
|
ff_ebur128_loudness_range(s->r128_out, &lra_out); |
824 |
|
|
ff_ebur128_loudness_global(s->r128_out, &i_out); |
825 |
|
|
ff_ebur128_relative_threshold(s->r128_out, &thresh_out); |
826 |
|
|
for (c = 0; c < s->channels; c++) { |
827 |
|
|
double tmp; |
828 |
|
|
ff_ebur128_sample_peak(s->r128_out, c, &tmp); |
829 |
|
|
if ((c == 0) || (tmp > tp_out)) |
830 |
|
|
tp_out = tmp; |
831 |
|
|
} |
832 |
|
|
|
833 |
|
|
switch(s->print_format) { |
834 |
|
|
case NONE: |
835 |
|
|
break; |
836 |
|
|
|
837 |
|
|
case JSON: |
838 |
|
|
av_log(ctx, AV_LOG_INFO, |
839 |
|
|
"\n{\n" |
840 |
|
|
"\t\"input_i\" : \"%.2f\",\n" |
841 |
|
|
"\t\"input_tp\" : \"%.2f\",\n" |
842 |
|
|
"\t\"input_lra\" : \"%.2f\",\n" |
843 |
|
|
"\t\"input_thresh\" : \"%.2f\",\n" |
844 |
|
|
"\t\"output_i\" : \"%.2f\",\n" |
845 |
|
|
"\t\"output_tp\" : \"%+.2f\",\n" |
846 |
|
|
"\t\"output_lra\" : \"%.2f\",\n" |
847 |
|
|
"\t\"output_thresh\" : \"%.2f\",\n" |
848 |
|
|
"\t\"normalization_type\" : \"%s\",\n" |
849 |
|
|
"\t\"target_offset\" : \"%.2f\"\n" |
850 |
|
|
"}\n", |
851 |
|
|
i_in, |
852 |
|
|
20. * log10(tp_in), |
853 |
|
|
lra_in, |
854 |
|
|
thresh_in, |
855 |
|
|
i_out, |
856 |
|
|
20. * log10(tp_out), |
857 |
|
|
lra_out, |
858 |
|
|
thresh_out, |
859 |
|
|
s->frame_type == LINEAR_MODE ? "linear" : "dynamic", |
860 |
|
|
s->target_i - i_out |
861 |
|
|
); |
862 |
|
|
break; |
863 |
|
|
|
864 |
|
|
case SUMMARY: |
865 |
|
|
av_log(ctx, AV_LOG_INFO, |
866 |
|
|
"\n" |
867 |
|
|
"Input Integrated: %+6.1f LUFS\n" |
868 |
|
|
"Input True Peak: %+6.1f dBTP\n" |
869 |
|
|
"Input LRA: %6.1f LU\n" |
870 |
|
|
"Input Threshold: %+6.1f LUFS\n" |
871 |
|
|
"\n" |
872 |
|
|
"Output Integrated: %+6.1f LUFS\n" |
873 |
|
|
"Output True Peak: %+6.1f dBTP\n" |
874 |
|
|
"Output LRA: %6.1f LU\n" |
875 |
|
|
"Output Threshold: %+6.1f LUFS\n" |
876 |
|
|
"\n" |
877 |
|
|
"Normalization Type: %s\n" |
878 |
|
|
"Target Offset: %+6.1f LU\n", |
879 |
|
|
i_in, |
880 |
|
|
20. * log10(tp_in), |
881 |
|
|
lra_in, |
882 |
|
|
thresh_in, |
883 |
|
|
i_out, |
884 |
|
|
20. * log10(tp_out), |
885 |
|
|
lra_out, |
886 |
|
|
thresh_out, |
887 |
|
|
s->frame_type == LINEAR_MODE ? "Linear" : "Dynamic", |
888 |
|
|
s->target_i - i_out |
889 |
|
|
); |
890 |
|
|
break; |
891 |
|
|
} |
892 |
|
|
|
893 |
|
|
end: |
894 |
|
|
if (s->r128_in) |
895 |
|
|
ff_ebur128_destroy(&s->r128_in); |
896 |
|
|
if (s->r128_out) |
897 |
|
|
ff_ebur128_destroy(&s->r128_out); |
898 |
|
|
av_freep(&s->limiter_buf); |
899 |
|
|
av_freep(&s->prev_smp); |
900 |
|
|
av_freep(&s->buf); |
901 |
|
|
} |
902 |
|
|
|
903 |
|
|
static const AVFilterPad avfilter_af_loudnorm_inputs[] = { |
904 |
|
|
{ |
905 |
|
|
.name = "default", |
906 |
|
|
.type = AVMEDIA_TYPE_AUDIO, |
907 |
|
|
.config_props = config_input, |
908 |
|
|
.filter_frame = filter_frame, |
909 |
|
|
}, |
910 |
|
|
{ NULL } |
911 |
|
|
}; |
912 |
|
|
|
913 |
|
|
static const AVFilterPad avfilter_af_loudnorm_outputs[] = { |
914 |
|
|
{ |
915 |
|
|
.name = "default", |
916 |
|
|
.request_frame = request_frame, |
917 |
|
|
.type = AVMEDIA_TYPE_AUDIO, |
918 |
|
|
}, |
919 |
|
|
{ NULL } |
920 |
|
|
}; |
921 |
|
|
|
922 |
|
|
AVFilter ff_af_loudnorm = { |
923 |
|
|
.name = "loudnorm", |
924 |
|
|
.description = NULL_IF_CONFIG_SMALL("EBU R128 loudness normalization"), |
925 |
|
|
.priv_size = sizeof(LoudNormContext), |
926 |
|
|
.priv_class = &loudnorm_class, |
927 |
|
|
.query_formats = query_formats, |
928 |
|
|
.init = init, |
929 |
|
|
.uninit = uninit, |
930 |
|
|
.inputs = avfilter_af_loudnorm_inputs, |
931 |
|
|
.outputs = avfilter_af_loudnorm_outputs, |
932 |
|
|
}; |