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/*
Copyright (C) 2021 Carl Hetherington <cth@carlh.net>
This file is part of DCP-o-matic.
DCP-o-matic is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
DCP-o-matic 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 General Public License for more details.
You should have received a copy of the GNU General Public License
along with DCP-o-matic. If not, see <http://www.gnu.org/licenses/>.
*/
#include "audio_analyser.h"
#include "audio_analysis.h"
#include "audio_buffers.h"
#include "audio_content.h"
#include "audio_filter_graph.h"
#include "audio_point.h"
#include "config.h"
#include "dcpomatic_log.h"
#include "film.h"
#include "filter.h"
#include "playlist.h"
#include <dcp/warnings.h>
extern "C" {
#include <leqm_nrt.h>
LIBDCP_DISABLE_WARNINGS
#include <libavutil/channel_layout.h>
#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
#include <libavfilter/f_ebur128.h>
#endif
LIBDCP_ENABLE_WARNINGS
}
using std::make_shared;
using std::max;
using std::shared_ptr;
using std::vector;
using namespace dcpomatic;
static auto constexpr num_points = 1024;
AudioAnalyser::AudioAnalyser(shared_ptr<const Film> film, shared_ptr<const Playlist> playlist, bool whole_film, std::function<void (float)> set_progress)
: _film(film)
, _playlist(playlist)
, _set_progress(set_progress)
#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
, _ebur128(film->audio_frame_rate(), film->audio_channels())
#endif
, _sample_peak(film->audio_channels())
, _sample_peak_frame(film->audio_channels())
, _analysis(film->audio_channels())
{
#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
_filters.push_back({"ebur128", "ebur128", "audio", "ebur128=peak=true"});
_ebur128.setup(_filters);
#endif
_current = std::vector<AudioPoint>(_film->audio_channels());
if (!whole_film) {
_start = _playlist->start().get_value_or(DCPTime());
}
for (int i = 0; i < film->audio_channels(); ++i) {
_sample_peak[i] = 0;
_sample_peak_frame[i] = 0;
}
auto add_if_required = [](vector<double>& v, size_t i, double db) {
if (v.size() > i) {
v[i] = pow(10, db / 20);
}
};
auto content = _playlist->content();
if (whole_film) {
_leqm_channels = film->audio_channels();
} else {
_leqm_channels = 0;
for (auto channel: content[0]->audio->mapping().mapped_output_channels()) {
/* This means that if, for example, a file only maps C we will
* calculate LEQ(m) for L, R and C. I'm not sure if this is
* right or not.
*/
_leqm_channels = std::min(film->audio_channels(), channel + 1);
}
}
/* XXX: is this right? Especially for more than 5.1? */
vector<double> channel_corrections(_leqm_channels, 1);
add_if_required(channel_corrections, 4, -3); // Ls
add_if_required(channel_corrections, 5, -3); // Rs
add_if_required(channel_corrections, 6, -144); // HI
add_if_required(channel_corrections, 7, -144); // VI
add_if_required(channel_corrections, 8, -3); // Lc
add_if_required(channel_corrections, 9, -3); // Rc
add_if_required(channel_corrections, 10, -3); // Lc
add_if_required(channel_corrections, 11, -3); // Rc
add_if_required(channel_corrections, 12, -144); // DBox
add_if_required(channel_corrections, 13, -144); // Sync
add_if_required(channel_corrections, 14, -144); // Sign Language
add_if_required(channel_corrections, 15, -144); // Unused
_leqm.reset(new leqm_nrt::Calculator(
_leqm_channels,
film->audio_frame_rate(),
24,
channel_corrections,
850, // suggested by leqm_nrt CLI source
64, // suggested by leqm_nrt CLI source
boost::thread::hardware_concurrency()
));
DCPTime const length = _playlist->length(_film);
Frame const len = DCPTime(length - _start).frames_round(film->audio_frame_rate());
_samples_per_point = max(int64_t(1), len / num_points);
}
void
AudioAnalyser::analyse(shared_ptr<AudioBuffers> b, DCPTime time)
{
LOG_DEBUG_AUDIO_ANALYSIS("AudioAnalyser received {} frames at {}", b->frames(), to_string(time));
DCPOMATIC_ASSERT(time >= _start);
/* In bug #2364 we had a lot of frames arriving here (~47s worth) which
* caused an OOM error on Windows. Check for the number of frames being
* reasonable here to make sure we catch this if it happens again.
*/
DCPOMATIC_ASSERT(b->frames() < 480000);
#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
if (Config::instance()->analyse_ebur128()) {
_ebur128.process(b);
}
#endif
int const frames = b->frames();
vector<double> interleaved(frames * _leqm_channels);
for (int j = 0; j < _leqm_channels; ++j) {
float const* data = b->data(j);
for (int i = 0; i < frames; ++i) {
float s = data[i];
interleaved[i * _leqm_channels + j] = s;
float as = fabsf(s);
if (as < 10e-7) {
/* We may struggle to serialise and recover inf or -inf, so prevent such
values by replacing with this (140dB down) */
s = as = 10e-7;
}
_current[j][AudioPoint::RMS] += pow(s, 2);
_current[j][AudioPoint::PEAK] = max(_current[j][AudioPoint::PEAK], as);
if (as > _sample_peak[j]) {
_sample_peak[j] = as;
_sample_peak_frame[j] = _done + i;
}
if (((_done + i) % _samples_per_point) == 0) {
_current[j][AudioPoint::RMS] = sqrt(_current[j][AudioPoint::RMS] / _samples_per_point);
_analysis.add_point(j, _current[j]);
_current[j] = AudioPoint();
}
}
}
_leqm->add(interleaved);
_done += frames;
DCPTime const length = _playlist->length(_film);
_set_progress((time.seconds() - _start.seconds()) / (length.seconds() - _start.seconds()));
LOG_DEBUG_AUDIO_ANALYSIS("Frames processed");
}
void
AudioAnalyser::finish()
{
vector<AudioAnalysis::PeakTime> sample_peak;
for (int i = 0; i < _film->audio_channels(); ++i) {
sample_peak.push_back(
AudioAnalysis::PeakTime(_sample_peak[i], DCPTime::from_frames(_sample_peak_frame[i], _film->audio_frame_rate()))
);
}
_analysis.set_sample_peak(sample_peak);
#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
if (Config::instance()->analyse_ebur128()) {
void* eb = _ebur128.get("Parsed_ebur128_0")->priv;
vector<float> true_peak;
for (int i = 0; i < _film->audio_channels(); ++i) {
true_peak.push_back(av_ebur128_get_true_peaks(eb)[i]);
}
_analysis.set_true_peak(true_peak);
_analysis.set_integrated_loudness(av_ebur128_get_integrated_loudness(eb));
_analysis.set_loudness_range(av_ebur128_get_loudness_range(eb));
}
#endif
if (_playlist->content().size() == 1) {
/* If there was only one piece of content in this analysis we may later need to know what its
gain was when we analysed it.
*/
if (auto ac = _playlist->content().front()->audio) {
_analysis.set_analysis_gain(ac->gain());
}
}
_analysis.set_samples_per_point(_samples_per_point);
_analysis.set_sample_rate(_film->audio_frame_rate());
_analysis.set_leqm(_leqm->leq_m());
}
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