Kyle Swanson
9 years ago
committed by
Paul B Mahol
parent
42ee137a0a
commit
c0c378009b
8 changed files with 972 additions and 1 deletions
@ -0,0 +1,907 @@ |
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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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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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ebur128_state *r128_in; |
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ebur128_state *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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{ "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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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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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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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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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; |
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} |
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for (n = 0; n < 30; n++) |
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s->delta[n] = pow(10., env_shortterm / 20.); |
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s->prev_delta = s->delta[s->index]; |
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s->buf_index = |
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s->limiter_buf_index = 0; |
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for (n = 0; n < (s->limiter_buf_size / inlink->channels); n++) { |
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for (c = 0; c < inlink->channels; c++) { |
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limiter_buf[s->limiter_buf_index + c] = buf[s->buf_index + c] * s->delta[s->index] * s->offset; |
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} |
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s->limiter_buf_index += inlink->channels; |
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if (s->limiter_buf_index >= s->limiter_buf_size) |
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s->limiter_buf_index -= s->limiter_buf_size; |
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s->buf_index += inlink->channels; |
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} |
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subframe_length = frame_size(inlink->sample_rate, 100); |
||||
true_peak_limiter(s, dst, subframe_length, inlink->channels); |
||||
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); |
||||
ebur128_add_frames_double(s->r128_out, dst, in->nb_samples); |
||||
|
||||
ebur128_loudness_range(s->r128_in, &lra); |
||||
ebur128_loudness_global(s->r128_in, &global); |
||||
ebur128_loudness_shortterm(s->r128_in, &shortterm); |
||||
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; |
||||
|
||||
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]; |
||||
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]; |
||||
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) |
||||
{ |
||||
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; |
||||
|
||||
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 = ebur128_init(inlink->channels, inlink->sample_rate, EBUR128_MODE_I | EBUR128_MODE_S | EBUR128_MODE_LRA | EBUR128_MODE_SAMPLE_PEAK); |
||||
if (!s->r128_in) |
||||
return AVERROR(ENOMEM); |
||||
|
||||
s->r128_out = ebur128_init(inlink->channels, inlink->sample_rate, EBUR128_MODE_I | EBUR128_MODE_S | EBUR128_MODE_LRA | EBUR128_MODE_SAMPLE_PEAK); |
||||
if (!s->r128_out) |
||||
return AVERROR(ENOMEM); |
||||
|
||||
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); |
||||
|
||||
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; |
||||
} |
||||
} |
||||
} |
||||
|
||||
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 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; |
||||
|
||||
ebur128_loudness_range(s->r128_in, &lra_in); |
||||
ebur128_loudness_global(s->r128_in, &i_in); |
||||
ebur128_relative_threshold(s->r128_in, &thresh_in); |
||||
for (c = 0; c < s->channels; c++) { |
||||
double tmp; |
||||
ebur128_sample_peak(s->r128_in, c, &tmp); |
||||
if ((c == 0) || (tmp > tp_in)) |
||||
tp_in = tmp; |
||||
} |
||||
|
||||
ebur128_loudness_range(s->r128_out, &lra_out); |
||||
ebur128_loudness_global(s->r128_out, &i_out); |
||||
ebur128_relative_threshold(s->r128_out, &thresh_out); |
||||
for (c = 0; c < s->channels; c++) { |
||||
double tmp; |
||||
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; |
||||
} |
||||
|
||||
ebur128_destroy(&s->r128_in); |
||||
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, |
||||
.uninit = uninit, |
||||
.inputs = avfilter_af_loudnorm_inputs, |
||||
.outputs = avfilter_af_loudnorm_outputs, |
||||
}; |
||||
Loading…
Reference in new issue