/*
- Copyright (C) 2012-2018 Carl Hetherington <cth@carlh.net>
+ Copyright (C) 2012-2021 Carl Hetherington <cth@carlh.net>
This file is part of DCP-o-matic.
*/
-#include "audio_analysis.h"
-#include "audio_buffers.h"
+
#include "analyse_audio_job.h"
-#include "audio_content.h"
+#include "audio_analysis.h"
#include "compose.hpp"
#include "dcpomatic_log.h"
#include "film.h"
+#include "filter.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 <iostream>
#include "i18n.h"
-using std::string;
-using std::vector;
+
+using std::cout;
+using std::dynamic_pointer_cast;
+using std::make_shared;
using std::max;
using std::min;
-using std::cout;
using std::shared_ptr;
-using std::dynamic_pointer_cast;
+using std::string;
+using std::vector;
using namespace dcpomatic;
#if BOOST_VERSION >= 106100
using namespace boost::placeholders;
#endif
-int const AnalyseAudioJob::_num_points = 1024;
-
-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)
+ , _analyser (film, playlist, from_zero, boost::bind(&Job::set_progress, this, _1, false))
, _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 ();
- for (auto i: _filters) {
- delete const_cast<Filter*> (i);
- }
- delete[] _current;
- delete[] _sample_peak;
- delete[] _sample_peak_frame;
}
+
string
AnalyseAudioJob::name () const
{
return _("Analysing audio");
}
+
string
AnalyseAudioJob::json_name () const
{
return N_("analyse_audio");
}
+
void
AnalyseAudioJob::run ()
{
LOG_DEBUG_AUDIO_ANALYSIS_NC("AnalyseAudioJob::run");
- shared_ptr<Player> player (new Player(_film, _playlist));
+ auto player = make_shared<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 ()));
+ player->Audio.connect (bind(&AudioAnalyser::analyse, &_analyser, _1, _2));
bool has_any_audio = false;
for (auto c: _playlist->content()) {
}
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));
+ player->seek (_analyser.start(), true);
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.
- */
- 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());
- _analysis->write (_path);
+ _analyser.finish ();
+ auto analysis = _analyser.get();
+ analysis.write (_path);
LOG_DEBUG_AUDIO_ANALYSIS_NC("Job finished");
set_progress (1);
set_state (FINISHED_OK);
}
-
-void
-AnalyseAudioJob::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");
-}
*/
+
/** @file src/lib/analyse_audio_job.h
* @brief AnalyseAudioJob class.
*/
+
+#include "audio_analyser.h"
#include "job.h"
#include "audio_point.h"
#include "types.h"
#include <leqm_nrt.h>
#include <boost/scoped_ptr.hpp>
+
class AudioBuffers;
class AudioAnalysis;
class Playlist;
class AudioFilterGraph;
class Filter;
+
/** @class AnalyseAudioJob
* @brief A job to analyse the audio of a film and make a note of its
* broad peak and RMS levels.
}
private:
- void analyse (std::shared_ptr<const AudioBuffers>, dcpomatic::DCPTime time);
+ AudioAnalyser _analyser;
std::shared_ptr<const Playlist> _playlist;
/** playlist's audio analysis path when the job was created */
boost::filesystem::path _path;
- dcpomatic::DCPTime _start;
- bool _from_zero;
-
- Frame _done;
- Frame _samples_per_point;
- AudioPoint* _current;
-
- float* _sample_peak;
- Frame* _sample_peak_frame;
-
- std::shared_ptr<AudioAnalysis> _analysis;
-
- std::shared_ptr<AudioFilterGraph> _ebur128;
- std::vector<Filter const *> _filters;
-
- boost::scoped_ptr<leqm_nrt::Calculator> _leqm;
static const int _num_points;
};
--- /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());
+}
--- /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_analysis.h"
+#include "dcpomatic_time.h"
+#include "types.h"
+#include <leqm_nrt.h>
+#include <boost/scoped_ptr.hpp>
+#include <memory>
+
+
+class AudioAnalysis;
+class AudioBuffers;
+class AudioFilterGraph;
+class AudioPoint;
+class Film;
+class Filter;
+class Playlist;
+
+
+class AudioAnalyser
+{
+public:
+ AudioAnalyser (std::shared_ptr<const Film> film, std::shared_ptr<const Playlist> playlist, bool from_zero, std::function<void (float)> set_progress);
+ ~AudioAnalyser ();
+
+ AudioAnalyser (AudioAnalyser const&) = delete;
+ AudioAnalyser& operator= (AudioAnalyser const&) = delete;
+
+ void analyse (std::shared_ptr<const AudioBuffers>, dcpomatic::DCPTime time);
+
+ dcpomatic::DCPTime start () const {
+ return _start;
+ }
+
+ void finish ();
+
+ AudioAnalysis get () const {
+ return _analysis;
+ }
+
+private:
+ std::shared_ptr<const Film> _film;
+ std::shared_ptr<const Playlist> _playlist;
+
+ std::function<void (float)> _set_progress;
+
+ dcpomatic::DCPTime _start;
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+ std::shared_ptr<AudioFilterGraph> _ebur128;
+#endif
+ std::vector<Filter const *> _filters;
+ Frame _samples_per_point = 1;
+
+ boost::scoped_ptr<leqm_nrt::Calculator> _leqm;
+ Frame _done = 0;
+ float* _sample_peak = nullptr;
+ Frame* _sample_peak_frame = nullptr;
+ AudioPoint* _current = nullptr;
+
+ AudioAnalysis _analysis;
+};
+
class Playlist;
-class AudioAnalysis : public boost::noncopyable
+class AudioAnalysis
{
public:
explicit AudioAnalysis (int c);
atmos_decoder.cc
atmos_metadata.cc
atmos_mxf_decoder.cc
+ audio_analyser.cc
audio_analysis.cc
audio_buffers.cc
audio_content.cc
projects / dcpomatic.git / commitdiff