Rename SafeStringStream -> locked_stringstream. Bump deps for removal of stringstream.
[dcpomatic.git] / src / lib / analyse_audio_job.cc
index 2146b03c58aa2dbd7a7fa3c825bfd29244382bb7..3e771d3f6af6a35c1e7a9e6a45b2be48b1b6b1ea 100644 (file)
@@ -1,30 +1,42 @@
 /*
     Copyright (C) 2012-2015 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 "film.h"
 #include "player.h"
 #include "playlist.h"
+#include "filter.h"
+#include "audio_filter_graph.h"
+#include "config.h"
+extern "C" {
+#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"
 
@@ -42,10 +54,25 @@ AnalyseAudioJob::AnalyseAudioJob (shared_ptr<const Film> film, shared_ptr<const
        , _playlist (playlist)
        , _done (0)
        , _samples_per_point (1)
-       , _overall_peak (0)
-       , _overall_peak_frame (0)
+       , _current (0)
+       , _sample_peak (0)
+       , _sample_peak_frame (0)
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+       , _ebur128 (new AudioFilterGraph (film->audio_frame_rate(), film->audio_channels()))
+#endif
 {
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+       _filters.push_back (new Filter ("ebur128", "ebur128", "audio", "ebur128=peak=true"));
+       _ebur128->setup (_filters);
+#endif
+}
 
+AnalyseAudioJob::~AnalyseAudioJob ()
+{
+       BOOST_FOREACH (Filter const * i, _filters) {
+               delete const_cast<Filter*> (i);
+       }
+       delete[] _current;
 }
 
 string
@@ -65,16 +92,22 @@ AnalyseAudioJob::run ()
 {
        shared_ptr<Player> player (new Player (_film, _playlist));
        player->set_ignore_video ();
+       player->set_fast ();
+       player->set_play_referenced ();
 
-       int64_t const len = _playlist->length().frames (_film->audio_frame_rate());
+       DCPTime const start = _playlist->start().get_value_or (DCPTime ());
+       DCPTime const length = _playlist->length ();
+
+       Frame const len = DCPTime (length - start).frames_round (_film->audio_frame_rate());
        _samples_per_point = max (int64_t (1), len / _num_points);
 
-       _current.resize (_film->audio_channels ());
+       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 (dynamic_pointer_cast<AudioContent> (c)) {
+               if (c->audio) {
                        has_any_audio = true;
                }
        }
@@ -82,14 +115,43 @@ AnalyseAudioJob::run ()
        if (has_any_audio) {
                _done = 0;
                DCPTime const block = DCPTime::from_seconds (1.0 / 8);
-               for (DCPTime t; t < _film->length(); t += block) {
-                       analyse (player->get_audio (t, block, false));
-                       set_progress (t.seconds() / _film->length().seconds());
+               for (DCPTime t = start; t < length; t += block) {
+                       shared_ptr<const AudioBuffers> audio = player->get_audio (t, block, false);
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+                       if (Config::instance()->analyse_ebur128 ()) {
+                               _ebur128->process (audio);
+                       }
+#endif
+                       analyse (audio);
+                       set_progress ((t.seconds() - start.seconds()) / (length.seconds() - start.seconds()));
                }
        }
 
-       _analysis->set_peak (_overall_peak, DCPTime::from_frames (_overall_peak_frame, _film->audio_frame_rate ()));
-       _analysis->write (_film->audio_analysis_path ());
+       _analysis->set_sample_peak (_sample_peak, DCPTime::from_frames (_sample_peak_frame, _film->audio_frame_rate ()));
+
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+       if (Config::instance()->analyse_ebur128 ()) {
+               void* eb = _ebur128->get("Parsed_ebur128_0")->priv;
+               double true_peak = 0;
+               for (int i = 0; i < _film->audio_channels(); ++i) {
+                       true_peak = max (true_peak, 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;
+               DCPOMATIC_ASSERT (ac);
+               _analysis->set_analysis_gain (ac->gain ());
+       }
+
+       _analysis->write (_film->audio_analysis_path (_playlist));
 
        set_progress (1);
        set_state (FINISHED_OK);
@@ -98,34 +160,34 @@ AnalyseAudioJob::run ()
 void
 AnalyseAudioJob::analyse (shared_ptr<const AudioBuffers> b)
 {
-       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) {
-                               /* SafeStringStream can't serialise and recover inf or -inf, so prevent such
+       int const frames = b->frames ();
+       int const channels = b->channels ();
+
+       for (int j = 0; j < channels; ++j) {
+               float* data = b->data(j);
+               for (int i = 0; i < frames; ++i) {
+                       float s = data[i];
+                       float as = fabsf (s);
+                       if (as < 10e-7) {
+                               /* locked_stringstream can't 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));
-
-                       float const as = fabs (s);
-
                        _current[j][AudioPoint::PEAK] = max (_current[j][AudioPoint::PEAK], as);
 
-                       if (as > _overall_peak) {
-                               _overall_peak = as;
-                               _overall_peak_frame = _done + i;
+                       if (as > _sample_peak) {
+                               _sample_peak = as;
+                               _sample_peak_frame = _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;
        }
+
+       _done += frames;
 }