/*
- Copyright (C) 2012 Carl Hetherington <cth@carlh.net>
+ Copyright (C) 2012-2018 Carl Hetherington <cth@carlh.net>
- This program is free software; you can redistribute it and/or modify
+ 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.
- This program is distributed in the hope that it will be useful,
+ 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 this program; if not, write to the Free Software
- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ along with DCP-o-matic. If not, see <http://www.gnu.org/licenses/>.
*/
#include "audio_analysis.h"
+#include "audio_buffers.h"
#include "analyse_audio_job.h"
+#include "audio_content.h"
#include "compose.hpp"
+#include "dcpomatic_log.h"
#include "film.h"
#include "player.h"
+#include "playlist.h"
+#include "filter.h"
+#include "audio_filter_graph.h"
+#include "config.h"
+extern "C" {
+#include <leqm_nrt.h>
+#include <libavutil/channel_layout.h>
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+#include <libavfilter/f_ebur128.h>
+#endif
+}
+#include <boost/foreach.hpp>
+#include <iostream>
#include "i18n.h"
using std::string;
+using std::vector;
using std::max;
using std::min;
using std::cout;
-using boost::shared_ptr;
+using std::shared_ptr;
+using std::dynamic_pointer_cast;
+using namespace dcpomatic;
+#if BOOST_VERSION >= 106100
+using namespace boost::placeholders;
+#endif
int const AnalyseAudioJob::_num_points = 1024;
-AnalyseAudioJob::AnalyseAudioJob (shared_ptr<Film> f)
- : Job (f)
+static void add_if_required(vector<double>& v, size_t i, double db)
+{
+ if (v.size() > i) {
+ v[i] = pow(10, db / 20);
+ }
+}
+
+/** @param from_zero true to analyse audio from time 0 in the playlist, otherwise begin at Playlist::start */
+AnalyseAudioJob::AnalyseAudioJob (shared_ptr<const Film> film, shared_ptr<const Playlist> playlist, bool from_zero)
+ : Job (film)
+ , _playlist (playlist)
+ , _path (film->audio_analysis_path(playlist))
+ , _from_zero (from_zero)
, _done (0)
, _samples_per_point (1)
+ , _current (0)
+ , _sample_peak (new float[film->audio_channels()])
+ , _sample_peak_frame (new Frame[film->audio_channels()])
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+ , _ebur128 (new AudioFilterGraph (film->audio_frame_rate(), film->audio_channels()))
+#endif
{
+ LOG_DEBUG_AUDIO_ANALYSIS_NC("AnalyseAudioJob::AnalyseAudioJob");
+
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+ _filters.push_back (new Filter ("ebur128", "ebur128", "audio", "ebur128=peak=true"));
+ _ebur128->setup (_filters);
+#endif
+
+ for (int i = 0; i < film->audio_channels(); ++i) {
+ _sample_peak[i] = 0;
+ _sample_peak_frame[i] = 0;
+ }
+
+ if (!_from_zero) {
+ _start = _playlist->start().get_value_or(DCPTime());
+ }
+
+ /* 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()
+ ));
+}
+AnalyseAudioJob::~AnalyseAudioJob ()
+{
+ stop_thread ();
+ BOOST_FOREACH (Filter const * i, _filters) {
+ delete const_cast<Filter*> (i);
+ }
+ delete[] _current;
+ delete[] _sample_peak;
+ delete[] _sample_peak_frame;
}
string
AnalyseAudioJob::name () const
{
- return String::compose (_("Analyse audio of %1"), _film->name());
+ return _("Analysing audio");
+}
+
+string
+AnalyseAudioJob::json_name () const
+{
+ return N_("analyse_audio");
}
void
AnalyseAudioJob::run ()
{
- shared_ptr<Player> player = _film->player ();
- player->disable_video ();
-
- player->Audio.connect (bind (&AnalyseAudioJob::audio, this, _1));
-
- _samples_per_point = max (int64_t (1), _film->audio_length() / _num_points);
-
- _current.resize (MAX_AUDIO_CHANNELS);
- _analysis.reset (new AudioAnalysis (MAX_AUDIO_CHANNELS));
-
- while (!player->pass()) {
- set_progress (float (_done) / _film->audio_length ());
+ LOG_DEBUG_AUDIO_ANALYSIS_NC("AnalyseAudioJob::run");
+
+ shared_ptr<Player> player (new Player(_film, _playlist));
+ player->set_ignore_video ();
+ player->set_ignore_text ();
+ player->set_fast ();
+ player->set_play_referenced ();
+ player->Audio.connect (bind (&AnalyseAudioJob::analyse, this, _1, _2));
+
+ 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);
+
+ delete[] _current;
+ _current = new AudioPoint[_film->audio_channels ()];
+ _analysis.reset (new AudioAnalysis (_film->audio_channels ()));
+
+ bool has_any_audio = false;
+ BOOST_FOREACH (shared_ptr<Content> c, _playlist->content ()) {
+ if (c->audio) {
+ has_any_audio = true;
+ }
}
- _analysis->write (_film->audio_analysis_path ());
-
+ if (has_any_audio) {
+ LOG_DEBUG_AUDIO_ANALYSIS("Seeking to %1", to_string(_start));
+ player->seek (_start, true);
+ _done = 0;
+ LOG_DEBUG_AUDIO_ANALYSIS("Starting loop for playlist of length %1", to_string(length));
+ while (!player->pass ()) {}
+ }
+
+ LOG_DEBUG_AUDIO_ANALYSIS_NC("Loop complete");
+
+ 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.
+ */
+ shared_ptr<const AudioContent> ac = _playlist->content().front()->audio;
+ if (ac) {
+ _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());
+ _analysis->write (_path);
+
+ LOG_DEBUG_AUDIO_ANALYSIS_NC("Job finished");
set_progress (1);
set_state (FINISHED_OK);
}
void
-AnalyseAudioJob::audio (shared_ptr<AudioBuffers> b)
+AnalyseAudioJob::analyse (shared_ptr<const AudioBuffers> b, DCPTime time)
{
- for (int i = 0; i < b->frames(); ++i) {
- for (int j = 0; j < b->channels(); ++j) {
- float s = b->data(j)[i];
- if (fabsf (s) < 10e-7) {
- /* stringstream can't serialise and recover inf or -inf, so prevent such
+ 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 = 10e-7;
+ s = as = 10e-7;
}
_current[j][AudioPoint::RMS] += pow (s, 2);
- _current[j][AudioPoint::PEAK] = max (_current[j][AudioPoint::PEAK], fabsf (s));
+ _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 % _samples_per_point) == 0) {
+ 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 ();
}
}
-
- ++_done;
}
-}
+ _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");
+}