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"
37 DCPOMATIC_DISABLE_WARNINGS
38 #include <libavutil/channel_layout.h>
39 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
40 #include <libavfilter/f_ebur128.h>
42 DCPOMATIC_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 (new float[film->audio_channels()])
64 , _sample_peak_frame (new 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 = new 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 ()
131 for (auto i: _filters) {
132 delete const_cast<Filter*> (i);
134 delete[] _sample_peak;
135 delete[] _sample_peak_frame;
140 AudioAnalyser::analyse (shared_ptr<const AudioBuffers> b, DCPTime time)
142 LOG_DEBUG_AUDIO_ANALYSIS("Received %1 frames at %2", b->frames(), to_string(time));
143 DCPOMATIC_ASSERT (time >= _start);
145 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
146 if (Config::instance()->analyse_ebur128 ()) {
147 _ebur128->process (b);
151 int const frames = b->frames ();
152 int const channels = b->channels ();
153 vector<double> interleaved(frames * channels);
155 for (int j = 0; j < channels; ++j) {
156 float* data = b->data(j);
157 for (int i = 0; i < frames; ++i) {
160 interleaved[i * channels + j] = s;
162 float as = fabsf (s);
164 /* We may struggle to serialise and recover inf or -inf, so prevent such
165 values by replacing with this (140dB down) */
168 _current[j][AudioPoint::RMS] += pow (s, 2);
169 _current[j][AudioPoint::PEAK] = max (_current[j][AudioPoint::PEAK], as);
171 if (as > _sample_peak[j]) {
172 _sample_peak[j] = as;
173 _sample_peak_frame[j] = _done + i;
176 if (((_done + i) % _samples_per_point) == 0) {
177 _current[j][AudioPoint::RMS] = sqrt (_current[j][AudioPoint::RMS] / _samples_per_point);
178 _analysis.add_point (j, _current[j]);
179 _current[j] = AudioPoint ();
184 _leqm->add(interleaved);
188 DCPTime const length = _playlist->length (_film);
189 _set_progress ((time.seconds() - _start.seconds()) / (length.seconds() - _start.seconds()));
190 LOG_DEBUG_AUDIO_ANALYSIS_NC("Frames processed");
195 AudioAnalyser::finish ()
197 vector<AudioAnalysis::PeakTime> sample_peak;
198 for (int i = 0; i < _film->audio_channels(); ++i) {
199 sample_peak.push_back (
200 AudioAnalysis::PeakTime (_sample_peak[i], DCPTime::from_frames (_sample_peak_frame[i], _film->audio_frame_rate ()))
203 _analysis.set_sample_peak (sample_peak);
205 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
206 if (Config::instance()->analyse_ebur128 ()) {
207 void* eb = _ebur128->get("Parsed_ebur128_0")->priv;
208 vector<float> true_peak;
209 for (int i = 0; i < _film->audio_channels(); ++i) {
210 true_peak.push_back (av_ebur128_get_true_peaks(eb)[i]);
212 _analysis.set_true_peak (true_peak);
213 _analysis.set_integrated_loudness (av_ebur128_get_integrated_loudness(eb));
214 _analysis.set_loudness_range (av_ebur128_get_loudness_range(eb));
218 if (_playlist->content().size() == 1) {
219 /* If there was only one piece of content in this analysis we may later need to know what its
220 gain was when we analysed it.
222 if (auto ac = _playlist->content().front()->audio) {
223 _analysis.set_analysis_gain (ac->gain());
227 _analysis.set_samples_per_point (_samples_per_point);
228 _analysis.set_sample_rate (_film->audio_frame_rate ());
229 _analysis.set_leqm (_leqm->leq_m());