--- /dev/null
+/*
+ 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 "types.h"
+extern "C" {
+#include <leqm_nrt.h>
+#include <libavutil/channel_layout.h>
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+#include <libavfilter/f_ebur128.h>
+#endif
+}
+
+
+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 from_zero, std::function<void (float)> set_progress)
+ : _film (film)
+ , _playlist (playlist)
+ , _set_progress (set_progress)
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+ , _ebur128 (new AudioFilterGraph(film->audio_frame_rate(), film->audio_channels()))
+#endif
+ , _sample_peak (new float[film->audio_channels()])
+ , _sample_peak_frame (new Frame[film->audio_channels()])
+ , _analysis (film->audio_channels())
+{
+
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+ _filters.push_back (new Filter("ebur128", "ebur128", "audio", "ebur128=peak=true"));
+ _ebur128->setup (_filters);
+#endif
+
+ _current = new AudioPoint[_film->audio_channels()];
+
+ if (!from_zero) {
+ _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);
+ }
+ };
+
+ /* XXX: is this right? Especially for more than 5.1? */
+ vector<double> channel_corrections(film->audio_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(
+ film->audio_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);
+}
+
+
+AudioAnalyser::~AudioAnalyser ()
+{
+ delete[] _current;
+ for (auto i: _filters) {
+ delete const_cast<Filter*> (i);
+ }
+ delete[] _sample_peak;
+ delete[] _sample_peak_frame;
+}
+
+
+void
+AudioAnalyser::analyse (shared_ptr<const AudioBuffers> b, DCPTime time)
+{
+ LOG_DEBUG_AUDIO_ANALYSIS("Received %1 frames at %2", b->frames(), to_string(time));
+ DCPOMATIC_ASSERT (time >= _start);
+
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+ if (Config::instance()->analyse_ebur128 ()) {
+ _ebur128->process (b);
+ }
+#endif
+
+ int const frames = b->frames ();
+ int const channels = b->channels ();
+ vector<double> interleaved(frames * channels);
+
+ for (int j = 0; j < channels; ++j) {
+ float* data = b->data(j);
+ for (int i = 0; i < frames; ++i) {
+ float s = data[i];
+
+ interleaved[i * 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_NC("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());
+}
projects / dcpomatic.git / blobdiff