2 Copyright (C) 2021 Carl Hetherington <cth@carlh.net>
4 This file is part of DCP-o-matic.
6 DCP-o-matic is free software; you can redistribute it and/or modify
7 it under the terms of the GNU General Public License as published by
8 the Free Software Foundation; either version 2 of the License, or
9 (at your option) any later version.
11 DCP-o-matic is distributed in the hope that it will be useful,
12 but WITHOUT ANY WARRANTY; without even the implied warranty of
13 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14 GNU General Public License for more details.
16 You should have received a copy of the GNU General Public License
17 along with DCP-o-matic. If not, see <http://www.gnu.org/licenses/>.
22 #include "audio_analyser.h"
23 #include "audio_analysis.h"
24 #include "audio_buffers.h"
25 #include "audio_content.h"
26 #include "audio_filter_graph.h"
27 #include "audio_point.h"
29 #include "dcpomatic_log.h"
34 #include <dcp/warnings.h>
37 LIBDCP_DISABLE_WARNINGS
38 #include <libavutil/channel_layout.h>
39 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
40 #include <libavfilter/f_ebur128.h>
42 LIBDCP_ENABLE_WARNINGS
46 using std::make_shared;
48 using std::shared_ptr;
50 using namespace dcpomatic;
53 static auto constexpr num_points = 1024;
56 AudioAnalyser::AudioAnalyser (shared_ptr<const Film> film, shared_ptr<const Playlist> playlist, bool from_zero, std::function<void (float)> set_progress)
58 , _playlist (playlist)
59 , _set_progress (set_progress)
60 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
61 , _ebur128 (new AudioFilterGraph(film->audio_frame_rate(), film->audio_channels()))
63 , _sample_peak (film->audio_channels())
64 , _sample_peak_frame (film->audio_channels())
65 , _analysis (film->audio_channels())
68 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
69 _filters.push_back (new Filter("ebur128", "ebur128", "audio", "ebur128=peak=true"));
70 _ebur128->setup (_filters);
73 _current = std::vector<AudioPoint>(_film->audio_channels());
76 _start = _playlist->start().get_value_or(DCPTime());
79 for (int i = 0; i < film->audio_channels(); ++i) {
81 _sample_peak_frame[i] = 0;
84 auto add_if_required = [](vector<double>& v, size_t i, double db) {
86 v[i] = pow(10, db / 20);
90 int leqm_channels = film->audio_channels();
91 auto content = _playlist->content();
92 if (content.size() == 1 && content[0]->audio) {
93 leqm_channels = content[0]->audio->mapping().mapped_output_channels().size();
96 /* XXX: is this right? Especially for more than 5.1? */
97 vector<double> channel_corrections(leqm_channels, 1);
98 add_if_required (channel_corrections, 4, -3); // Ls
99 add_if_required (channel_corrections, 5, -3); // Rs
100 add_if_required (channel_corrections, 6, -144); // HI
101 add_if_required (channel_corrections, 7, -144); // VI
102 add_if_required (channel_corrections, 8, -3); // Lc
103 add_if_required (channel_corrections, 9, -3); // Rc
104 add_if_required (channel_corrections, 10, -3); // Lc
105 add_if_required (channel_corrections, 11, -3); // Rc
106 add_if_required (channel_corrections, 12, -144); // DBox
107 add_if_required (channel_corrections, 13, -144); // Sync
108 add_if_required (channel_corrections, 14, -144); // Sign Language
109 add_if_required (channel_corrections, 15, -144); // Unused
111 _leqm.reset(new leqm_nrt::Calculator(
113 film->audio_frame_rate(),
116 850, // suggested by leqm_nrt CLI source
117 64, // suggested by leqm_nrt CLI source
118 boost::thread::hardware_concurrency()
121 DCPTime const length = _playlist->length (_film);
123 Frame const len = DCPTime (length - _start).frames_round (film->audio_frame_rate());
124 _samples_per_point = max (int64_t (1), len / num_points);
128 AudioAnalyser::~AudioAnalyser ()
130 for (auto i: _filters) {
131 delete const_cast<Filter*> (i);
137 AudioAnalyser::analyse (shared_ptr<AudioBuffers> b, DCPTime time)
139 LOG_DEBUG_AUDIO_ANALYSIS("Received %1 frames at %2", b->frames(), to_string(time));
140 DCPOMATIC_ASSERT (time >= _start);
142 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
143 if (Config::instance()->analyse_ebur128 ()) {
144 _ebur128->process (b);
148 int const frames = b->frames ();
149 int const channels = b->channels ();
150 vector<double> interleaved(frames * channels);
152 for (int j = 0; j < channels; ++j) {
153 float const* data = b->data(j);
154 for (int i = 0; i < frames; ++i) {
157 interleaved[i * channels + j] = s;
159 float as = fabsf (s);
161 /* We may struggle to serialise and recover inf or -inf, so prevent such
162 values by replacing with this (140dB down) */
165 _current[j][AudioPoint::RMS] += pow (s, 2);
166 _current[j][AudioPoint::PEAK] = max (_current[j][AudioPoint::PEAK], as);
168 if (as > _sample_peak[j]) {
169 _sample_peak[j] = as;
170 _sample_peak_frame[j] = _done + i;
173 if (((_done + i) % _samples_per_point) == 0) {
174 _current[j][AudioPoint::RMS] = sqrt (_current[j][AudioPoint::RMS] / _samples_per_point);
175 _analysis.add_point (j, _current[j]);
176 _current[j] = AudioPoint ();
181 _leqm->add(interleaved);
185 DCPTime const length = _playlist->length (_film);
186 _set_progress ((time.seconds() - _start.seconds()) / (length.seconds() - _start.seconds()));
187 LOG_DEBUG_AUDIO_ANALYSIS_NC("Frames processed");
192 AudioAnalyser::finish ()
194 vector<AudioAnalysis::PeakTime> sample_peak;
195 for (int i = 0; i < _film->audio_channels(); ++i) {
196 sample_peak.push_back (
197 AudioAnalysis::PeakTime (_sample_peak[i], DCPTime::from_frames (_sample_peak_frame[i], _film->audio_frame_rate ()))
200 _analysis.set_sample_peak (sample_peak);
202 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
203 if (Config::instance()->analyse_ebur128 ()) {
204 void* eb = _ebur128->get("Parsed_ebur128_0")->priv;
205 vector<float> true_peak;
206 for (int i = 0; i < _film->audio_channels(); ++i) {
207 true_peak.push_back (av_ebur128_get_true_peaks(eb)[i]);
209 _analysis.set_true_peak (true_peak);
210 _analysis.set_integrated_loudness (av_ebur128_get_integrated_loudness(eb));
211 _analysis.set_loudness_range (av_ebur128_get_loudness_range(eb));
215 if (_playlist->content().size() == 1) {
216 /* If there was only one piece of content in this analysis we may later need to know what its
217 gain was when we analysed it.
219 if (auto ac = _playlist->content().front()->audio) {
220 _analysis.set_analysis_gain (ac->gain());
224 _analysis.set_samples_per_point (_samples_per_point);
225 _analysis.set_sample_rate (_film->audio_frame_rate ());
226 _analysis.set_leqm (_leqm->leq_m());