diff options
| -rw-r--r-- | Changelog | 1 | ||||
| -rw-r--r-- | MAINTAINERS | 1 | ||||
| -rwxr-xr-x | configure | 5 | ||||
| -rw-r--r-- | doc/filters.texi | 55 | ||||
| -rw-r--r-- | libavfilter/Makefile | 1 | ||||
| -rw-r--r-- | libavfilter/af_loudnorm.c | 907 | ||||
| -rw-r--r-- | libavfilter/allfilters.c | 1 | ||||
| -rw-r--r-- | libavfilter/version.h | 2 |
8 files changed, 972 insertions, 1 deletions
@@ -35,6 +35,7 @@ version <next>: - Generic OpenMAX IL encoder with support for Raspberry Pi - IFF ANIM demuxer & decoder - Direct Stream Transfer (DST) decoder +- loudnorm filter version 3.0: - Common Encryption (CENC) MP4 encoding and decoding support diff --git a/MAINTAINERS b/MAINTAINERS index 14bf377d1f..52c30edba4 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -358,6 +358,7 @@ Filters: af_compand.c Paul B Mahol af_firequalizer.c Muhammad Faiz af_ladspa.c Paul B Mahol + af_loudnorm.c Kyle Swanson af_pan.c Nicolas George af_sidechaincompress.c Paul B Mahol af_silenceremove.c Paul B Mahol @@ -226,6 +226,8 @@ External library support: --enable-libcdio enable audio CD grabbing with libcdio [no] --enable-libdc1394 enable IIDC-1394 grabbing using libdc1394 and libraw1394 [no] + --enable-libebur128 enable libebur128 for EBU R128 measurement, + needed for loudnorm filter [no] --enable-libfaac enable AAC encoding via libfaac [no] --enable-libfdk-aac enable AAC de/encoding via libfdk-aac [no] --enable-libflite enable flite (voice synthesis) support via libflite [no] @@ -1472,6 +1474,7 @@ EXTERNAL_LIBRARY_LIST=" libcdio libcelt libdc1394 + libebur128 libfaac libfdk_aac libflite @@ -2987,6 +2990,7 @@ hqdn3d_filter_deps="gpl" interlace_filter_deps="gpl" kerndeint_filter_deps="gpl" ladspa_filter_deps="ladspa dlopen" +loudnorm_filter_deps="libebur128" mcdeint_filter_deps="avcodec gpl" movie_filter_deps="avcodec avformat" mpdecimate_filter_deps="gpl" @@ -5593,6 +5597,7 @@ enabled libcelt && require libcelt celt/celt.h celt_decode -lcelt0 && { check_lib celt/celt.h celt_decoder_create_custom -lcelt0 || die "ERROR: libcelt must be installed and version must be >= 0.11.0."; } enabled libcaca && require_pkg_config caca caca.h caca_create_canvas +enabled libebur128 && require ebur128 ebur128.h ebur128_relative_threshold -lebur128 enabled libfaac && require2 libfaac "stdint.h faac.h" faacEncGetVersion -lfaac enabled libfdk_aac && { use_pkg_config fdk-aac "fdk-aac/aacenc_lib.h" aacEncOpen || { require libfdk_aac fdk-aac/aacenc_lib.h aacEncOpen -lfdk-aac && diff --git a/doc/filters.texi b/doc/filters.texi index a7c480e68c..27584e91db 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -2711,6 +2711,61 @@ Modify the @var{N}-th control value. If the specified value is not valid, it is ignored and prior one is kept. @end table +@section loudnorm + +EBU R128 loudness normalization. Includes both dynamic and linear normalization modes. +Support for both single pass (livestreams, files) and double pass (files) modes. +This algorithm can target IL, LRA, and maximum true peak. + +To enable compilation of this filter you need to configure FFmpeg with +@code{--enable-libebur128}. + +The filter accepts the following options: + +@table @option +@item I, i +Set integrated loudness target. +Range is -70.0 - -5.0. Default value is -24.0. + +@item LRA, lra +Set loudness range target. +Range is 1.0 - 20.0. Default value is 7.0. + +@item TP, tp +Set maximum true peak. +Range is -9.0 - +0.0. Default value is -2.0. + +@item measured_I, measured_i +Measured IL of input file. +Range is -99.0 - +0.0. + +@item measured_LRA, measured_lra +Measured LRA of input file. +Range is 0.0 - 99.0. + +@item measured_TP, measured_tp +Measured true peak of input file. +Range is -99.0 - +99.0. + +@item measured_thresh +Measured threshold of input file. +Range is -99.0 - +0.0. + +@item offset +Set offset gain. Gain is applied before the true-peak limiter. +Range is -99.0 - +99.0. Default is +0.0. + +@item linear +Normalize linearly if possible. +measured_I, measured_LRA, measured_TP, and measured_thresh must also +to be specified in order to use this mode. +Options are true or false. Default is true. + +@item print_format +Set print format for stats. Options are summary, json, or none. +Default value is none. +@end table + @section lowpass Apply a low-pass filter with 3dB point frequency. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index aac2f573ba..65a831e526 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -89,6 +89,7 @@ OBJS-$(CONFIG_FLANGER_FILTER) += af_flanger.o generate_wave_table OBJS-$(CONFIG_HIGHPASS_FILTER) += af_biquads.o OBJS-$(CONFIG_JOIN_FILTER) += af_join.o OBJS-$(CONFIG_LADSPA_FILTER) += af_ladspa.o +OBJS-$(CONFIG_LOUDNORM_FILTER) += af_loudnorm.o OBJS-$(CONFIG_LOWPASS_FILTER) += af_biquads.o OBJS-$(CONFIG_PAN_FILTER) += af_pan.o OBJS-$(CONFIG_REPLAYGAIN_FILTER) += af_replaygain.o diff --git a/libavfilter/af_loudnorm.c b/libavfilter/af_loudnorm.c new file mode 100644 index 0000000000..cb210d41fe --- /dev/null +++ b/libavfilter/af_loudnorm.c @@ -0,0 +1,907 @@ +/* + * Copyright (c) 2016 Kyle Swanson <k@ylo.ph>. + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/* http://k.ylo.ph/2016/04/04/loudnorm.html */ + +#include "libavutil/opt.h" +#include "avfilter.h" +#include "internal.h" +#include "audio.h" +#include <ebur128.h> + +enum FrameType { + FIRST_FRAME, + INNER_FRAME, + FINAL_FRAME, + LINEAR_MODE, + FRAME_NB +}; + +enum LimiterState { + OUT, + ATTACK, + SUSTAIN, + RELEASE, + STATE_NB +}; + +enum PrintFormat { + NONE, + JSON, + SUMMARY, + PF_NB +}; + +typedef struct LoudNormContext { + const AVClass *class; + double target_i; + double target_lra; + double target_tp; + double measured_i; + double measured_lra; + double measured_tp; + double measured_thresh; + double offset; + int linear; + enum PrintFormat print_format; + + double *buf; + int buf_size; + int buf_index; + int prev_buf_index; + + double delta[30]; + double weights[21]; + double prev_delta; + int index; + + double gain_reduction[2]; + double *limiter_buf; + double *prev_smp; + int limiter_buf_index; + int limiter_buf_size; + enum LimiterState limiter_state; + int peak_index; + int env_index; + int env_cnt; + int attack_length; + int release_length; + + int64_t pts; + enum FrameType frame_type; + int above_threshold; + int prev_nb_samples; + int channels; + + ebur128_state *r128_in; + ebur128_state *r128_out; +} LoudNormContext; + +#define OFFSET(x) offsetof(LoudNormContext, x) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption loudnorm_options[] = { + { "I", "set integrated loudness target", OFFSET(target_i), AV_OPT_TYPE_DOUBLE, {.dbl = -24.}, -70., -5., FLAGS }, + { "i", "set integrated loudness target", OFFSET(target_i), AV_OPT_TYPE_DOUBLE, {.dbl = -24.}, -70., -5., FLAGS }, + { "LRA", "set loudness range target", OFFSET(target_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 7.}, 1., 20., FLAGS }, + { "lra", "set loudness range target", OFFSET(target_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 7.}, 1., 20., FLAGS }, + { "TP", "set maximum true peak", OFFSET(target_tp), AV_OPT_TYPE_DOUBLE, {.dbl = -2.}, -9., 0., FLAGS }, + { "tp", "set maximum true peak", OFFSET(target_tp), AV_OPT_TYPE_DOUBLE, {.dbl = -2.}, -9., 0., FLAGS }, + { "measured_I", "measured IL of input file", OFFSET(measured_i), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 0., FLAGS }, + { "measured_i", "measured IL of input file", OFFSET(measured_i), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 0., FLAGS }, + { "measured_LRA", "measured LRA of input file", OFFSET(measured_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, 0., 99., FLAGS }, + { "measured_lra", "measured LRA of input file", OFFSET(measured_lra), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, 0., 99., FLAGS }, + { "measured_TP", "measured true peak of input file", OFFSET(measured_tp), AV_OPT_TYPE_DOUBLE, {.dbl = 99.}, -99., 99., FLAGS }, + { "measured_tp", "measured true peak of input file", OFFSET(measured_tp), AV_OPT_TYPE_DOUBLE, {.dbl = 99.}, -99., 99., FLAGS }, + { "measured_thresh", "measured threshold of input file", OFFSET(measured_thresh), AV_OPT_TYPE_DOUBLE, {.dbl = -70.}, -99., 0., FLAGS }, + { "offset", "set offset gain", OFFSET(offset), AV_OPT_TYPE_DOUBLE, {.dbl = 0.}, -99., 99., FLAGS }, + { "linear", "normalize linearly if possible", OFFSET(linear), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS }, + { "print_format", "set print format for stats", OFFSET(print_format), AV_OPT_TYPE_INT, {.i64 = NONE}, NONE, PF_NB -1, FLAGS, "print_format" }, + { "none", 0, 0, AV_OPT_TYPE_CONST, {.i64 = NONE}, 0, 0, FLAGS, "print_format" }, + { "json", 0, 0, AV_OPT_TYPE_CONST, {.i64 = JSON}, 0, 0, FLAGS, "print_format" }, + { "summary", 0, 0, AV_OPT_TYPE_CONST, {.i64 = SUMMARY}, 0, 0, FLAGS, "print_format" }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(loudnorm); + +static inline int frame_size(int sample_rate, int frame_len_msec) +{ + const int frame_size = round((double)sample_rate * (frame_len_msec / 1000.0)); + return frame_size + (frame_size % 2); +} + +static void init_gaussian_filter(LoudNormContext *s) +{ + double total_weight = 0.0; + const double sigma = 3.5; + double adjust; + int i; + + const int offset = 21 / 2; + const double c1 = 1.0 / (sigma * sqrt(2.0 * M_PI)); + const double c2 = 2.0 * pow(sigma, 2.0); + + for (i = 0; i < 21; i++) { + const int x = i - offset; + s->weights[i] = c1 * exp(-(pow(x, 2.0) / c2)); + total_weight += s->weights[i]; + } + + adjust = 1.0 / total_weight; + for (i = 0; i < 21; i++) + s->weights[i] *= adjust; +} + +static double gaussian_filter(LoudNormContext *s, int index) +{ + double result = 0.; + int i; + + index = index - 10 > 0 ? index - 10 : index + 20; + for (i = 0; i < 21; i++) + result += s->delta[((index + i) < 30) ? (index + i) : (index + i - 30)] * s->weights[i]; + + return result; +} + +static void detect_peak(LoudNormContext *s, int offset, int nb_samples, int channels, int *peak_delta, double *peak_value) +{ + int n, c, i, index; + double ceiling; + double *buf; + + *peak_delta = -1; + buf = s->limiter_buf; + ceiling = s->target_tp; + + index = s->limiter_buf_index + (offset * channels) + (1920 * channels); + if (index >= s->limiter_buf_size) + index -= s->limiter_buf_size; + + if (s->frame_type == FIRST_FRAME) { + for (c = 0; c < channels; c++) + s->prev_smp[c] = fabs(buf[index + c - channels]); + } + + for (n = 0; n < nb_samples; n++) { + for (c = 0; c < channels; c++) { + double this, next, max_peak; + + this = fabs(buf[(index + c) < s->limiter_buf_size ? (index + c) : (index + c - s->limiter_buf_size)]); + next = fabs(buf[(index + c + channels) < s->limiter_buf_size ? (index + c + channels) : (index + c + channels - s->limiter_buf_size)]); + + if ((s->prev_smp[c] <= this) && (next <= this) && (this > ceiling) && (n > 0)) { + int detected; + + detected = 1; + for (i = 2; i < 12; i++) { + next = fabs(buf[(index + c + (i * channels)) < s->limiter_buf_size ? (index + c + (i * channels)) : (index + c + (i * channels) - s->limiter_buf_size)]); + if (next > this) { + detected = 0; + break; + } + } + + if (!detected) + continue; + + for (c = 0; c < channels; c++) { + if (c == 0 || fabs(buf[index + c]) > max_peak) + max_peak = fabs(buf[index + c]); + + s->prev_smp[c] = fabs(buf[(index + c) < s->limiter_buf_size ? (index + c) : (index + c - s->limiter_buf_size)]); + } + + *peak_delta = n; + s->peak_index = index; + *peak_value = max_peak; + return; + } + + s->prev_smp[c] = this; + } + + index += channels; + if (index >= s->limiter_buf_size) + index -= s->limiter_buf_size; + } +} + +static void true_peak_limiter(LoudNormContext *s, double *out, int nb_samples, int channels) +{ + int n, c, index, peak_delta, smp_cnt; + double ceiling, peak_value; + double *buf; + + buf = s->limiter_buf; + ceiling = s->target_tp; + index = s->limiter_buf_index; + smp_cnt = 0; + + if (s->frame_type == FIRST_FRAME) { + double max; + + max = 0.; + for (n = 0; n < 1920; n++) { + for (c = 0; c < channels; c++) { + max = fabs(buf[c]) > max ? fabs(buf[c]) : max; + } + buf += channels; + } + + if (max > ceiling) { + s->gain_reduction[1] = ceiling / max; + s->limiter_state = SUSTAIN; + buf = s->limiter_buf; + + for (n = 0; n < 1920; n++) { + for (c = 0; c < channels; c++) { + double env; + env = s->gain_reduction[1]; + buf[c] *= env; + } + buf += channels; + } + } + + buf = s->limiter_buf; + } + + do { + + switch(s->limiter_state) { + case OUT: + detect_peak(s, smp_cnt, nb_samples - smp_cnt, channels, &peak_delta, &peak_value); + if (peak_delta != -1) { + s->env_cnt = 0; + smp_cnt += (peak_delta - s->attack_length); + s->gain_reduction[0] = 1.; + s->gain_reduction[1] = ceiling / peak_value; + s->limiter_state = ATTACK; + + s->env_index = s->peak_index - (s->attack_length * channels); + if (s->env_index < 0) + s->env_index += s->limiter_buf_size; + + s->env_index += (s->env_cnt * channels); + if (s->env_index > s->limiter_buf_size) + s->env_index -= s->limiter_buf_size; + + } else { + smp_cnt = nb_samples; + } + break; + + case ATTACK: + for (; s->env_cnt < s->attack_length; s->env_cnt++) { + for (c = 0; c < channels; c++) { + double env; + env = s->gain_reduction[0] - ((double) s->env_cnt / (s->attack_length - 1) * (s->gain_reduction[0] - s->gain_reduction[1])); + buf[s->env_index + c] *= env; + } + + s->env_index += channels; + if (s->env_index >= s->limiter_buf_size) + s->env_index -= s->limiter_buf_size; + + smp_cnt++; + if (smp_cnt >= nb_samples) { + s->env_cnt++; + break; + } + } + + if (smp_cnt < nb_samples) { + s->env_cnt = 0; + s->attack_length = 1920; + s->limiter_state = SUSTAIN; + } + break; + + case SUSTAIN: + detect_peak(s, smp_cnt, nb_samples, channels, &peak_delta, &peak_value); + if (peak_delta == -1) { + s->limiter_state = RELEASE; + s->gain_reduction[0] = s->gain_reduction[1]; + s->gain_reduction[1] = 1.; + s->env_cnt = 0; + break; + } else { + double gain_reduction; + gain_reduction = ceiling / peak_value; + + if (gain_reduction < s->gain_reduction[1]) { + s->limiter_state = ATTACK; + + s->attack_length = peak_delta; + if (s->attack_length <= 1) + s->attack_length = 2; + + s->gain_reduction[0] = s->gain_reduction[1]; + s->gain_reduction[1] = gain_reduction; + s->env_cnt = 0; + break; + } + + for (s->env_cnt = 0; s->env_cnt < peak_delta; s->env_cnt++) { + for (c = 0; c < channels; c++) { + double env; + env = s->gain_reduction[1]; + buf[s->env_index + c] *= env; + } + + s->env_index += channels; + if (s->env_index >= s->limiter_buf_size) + s->env_index -= s->limiter_buf_size; + + smp_cnt++; + if (smp_cnt >= nb_samples) { + s->env_cnt++; + break; + } + } + } + break; + + case RELEASE: + for (; s->env_cnt < s->release_length; s->env_cnt++) { + for (c = 0; c < channels; c++) { + double env; + env = s->gain_reduction[0] + (((double) s->env_cnt / (s->release_length - 1)) * (s->gain_reduction[1] - s->gain_reduction[0])); + buf[s->env_index + c] *= env; + } + + s->env_index += channels; + if (s->env_index >= s->limiter_buf_size) + s->env_index -= s->limiter_buf_size; + + smp_cnt++; + if (smp_cnt >= nb_samples) { + s->env_cnt++; + break; + } + } + + if (smp_cnt < nb_samples) { + s->env_cnt = 0; + s->limiter_state = OUT; + } + + break; + } + + } while (smp_cnt < nb_samples); + + for (n = 0; n < nb_samples; n++) { + for (c = 0; c < channels; c++) { + out[c] = buf[index + c]; + if (fabs(out[c]) > ceiling) { + out[c] = ceiling * (out[c] < 0 ? -1 : 1); + } + } + out += channels; + index += channels; + if (index >= s->limiter_buf_size) + index -= s->limiter_buf_size; + } +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + LoudNormContext *s = ctx->priv; + AVFilterLink *outlink = ctx->outputs[0]; + AVFrame *out; + const double *src; + double *dst; + double *buf; + double *limiter_buf; + int i, n, c, subframe_length, src_index; + double gain, gain_next, env_global, env_shortterm, + global, shortterm, lra, relative_threshold; + + if (av_frame_is_writable(in)) { + out = in; + } else { + out = ff_get_audio_buffer(inlink, in->nb_samples); + if (!out) { + av_frame_free(&in); + return AVERROR(ENOMEM); + } + av_frame_copy_props(out, in); + } + + out->pts = s->pts; + src = (const double *)in->data[0]; + dst = (double *)out->data[0]; + buf = s->buf; + limiter_buf = s->limiter_buf; + + ebur128_add_frames_double(s->r128_in, src, in->nb_samples); + + if (s->frame_type == FIRST_FRAME && in->nb_samples < frame_size(inlink->sample_rate, 3000)) { + double offset, offset_tp, true_peak; + + ebur128_loudness_global(s->r128_in, &global); + for (c = 0; c < inlink->channels; c++) { + double tmp; + ebur128_sample_peak(s->r128_in, c, &tmp); + if (c == 0 || tmp > true_peak) + true_peak = tmp; + } + + offset = s->target_i - global; + offset_tp = true_peak + offset; + s->offset = offset_tp < s->target_tp ? offset : s->target_tp - true_peak; + s->offset = pow(10., s->offset / 20.); + s->frame_type = LINEAR_MODE; + } + + switch (s->frame_type) { + case FIRST_FRAME: + for (n = 0; n < in->nb_samples; n++) { + for (c = 0; c < inlink->channels; c++) { + buf[s->buf_index + c] = src[c]; + } + src += inlink->channels; + s->buf_index += inlink->channels; + } + + ebur128_loudness_shortterm(s->r128_in, &shortterm); + + if (shortterm < s->measured_thresh) { + s->above_threshold = 0; + env_shortterm = shortterm <= -70. ? 0. : s->target_i - s->measured_i; + } else { + s->above_threshold = 1; + env_shortterm = shortterm <= -70. ? 0. : s->target_i - shortterm; + } + + for (n = 0; n < 30; n++) + s->delta[n] = pow(10., env_shortterm / 20.); + s->prev_delta = s->delta[s->index]; + + s->buf_index = + s->limiter_buf_index = 0; + + for (n = 0; n < (s->limiter_buf_size / inlink->channels); n++) { + for (c = 0; c < inlink->channels; c++) { + limiter_buf[s->limiter_buf_index + c] = buf[s->buf_index + c] * s->delta[s->index] * s->offset; + } + s->limiter_buf_index += inlink->channels; + if (s->limiter_buf_index >= s->limiter_buf_size) + s->limiter_buf_index -= s->limiter_buf_size; + + s->buf_index += inlink->channels; + } + + subframe_length = frame_size(inlink->sample_rate, 100); + true_peak_limiter(s, dst, subframe_length, inlink->channels); + 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, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index a9725760fa..d0d491e4d5 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -108,6 +108,7 @@ void avfilter_register_all(void) REGISTER_FILTER(HIGHPASS, highpass, af); REGISTER_FILTER(JOIN, join, af); REGISTER_FILTER(LADSPA, ladspa, af); + REGISTER_FILTER(LOUDNORM, loudnorm, af); REGISTER_FILTER(LOWPASS, lowpass, af); REGISTER_FILTER(PAN, pan, af); REGISTER_FILTER(REPLAYGAIN, replaygain, af); diff --git a/libavfilter/version.h b/libavfilter/version.h index 1a9c4ac5cf..d693d6d5ec 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -30,7 +30,7 @@ #include "libavutil/version.h" #define LIBAVFILTER_VERSION_MAJOR 6 -#define LIBAVFILTER_VERSION_MINOR 45 +#define LIBAVFILTER_VERSION_MINOR 46 #define LIBAVFILTER_VERSION_MICRO 100 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \ |