Basic grunt-work, untested and unfinished, but it compiles.
[dcpomatic.git] / src / lib / analyse_audio_job.cc
index 769f3762bee98ee214c9972ddb0c6018c0cc8767..1378b66a433fc0fa5a531d4db682f4f26ea3de31 100644 (file)
@@ -1,19 +1,20 @@
 /*
     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 "playlist.h"
 #include "filter.h"
 #include "audio_filter_graph.h"
+#include "config.h"
 extern "C" {
 #include <libavutil/channel_layout.h>
-#ifdef DCPOMATIC_HAVE_PATCHED_FFMPEG
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
 #include <libavfilter/f_ebur128.h>
 #endif
 }
@@ -39,6 +41,7 @@ extern "C" {
 #include "i18n.h"
 
 using std::string;
+using std::vector;
 using std::max;
 using std::min;
 using std::cout;
@@ -53,16 +56,21 @@ AnalyseAudioJob::AnalyseAudioJob (shared_ptr<const Film> film, shared_ptr<const
        , _done (0)
        , _samples_per_point (1)
        , _current (0)
-       , _sample_peak (0)
-       , _sample_peak_frame (0)
-#ifdef DCPOMATIC_HAVE_PATCHED_FFMPEG
-       , _ebur128 (new AudioFilterGraph (film->audio_frame_rate(), av_get_default_channel_layout(film->audio_channels())))
+       , _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
 {
-#ifdef DCPOMATIC_HAVE_PATCHED_FFMPEG
+#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;
+       }
 }
 
 AnalyseAudioJob::~AnalyseAudioJob ()
@@ -71,6 +79,8 @@ AnalyseAudioJob::~AnalyseAudioJob ()
                delete const_cast<Filter*> (i);
        }
        delete[] _current;
+       delete[] _sample_peak;
+       delete[] _sample_peak_frame;
 }
 
 string
@@ -92,6 +102,7 @@ AnalyseAudioJob::run ()
        player->set_ignore_video ();
        player->set_fast ();
        player->set_play_referenced ();
+       player->Audio.connect (bind (&AnalyseAudioJob::analyse, this, _1, _2));
 
        DCPTime const start = _playlist->start().get_value_or (DCPTime ());
        DCPTime const length = _playlist->length ();
@@ -105,44 +116,44 @@ AnalyseAudioJob::run ()
 
        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;
                }
        }
 
        if (has_any_audio) {
                _done = 0;
-               DCPTime const block = DCPTime::from_seconds (1.0 / 8);
-               for (DCPTime t = start; t < length; t += block) {
-                       shared_ptr<const AudioBuffers> audio = player->get_audio (t, block, false);
-#ifdef DCPOMATIC_HAVE_PATCHED_FFMPEG
-                       _ebur128->process (audio);
-#endif
-                       analyse (audio);
-                       set_progress ((t.seconds() - start.seconds()) / (length.seconds() - start.seconds()));
-               }
+               while (!player->pass ()) {}
        }
 
-       _analysis->set_sample_peak (_sample_peak, DCPTime::from_frames (_sample_peak_frame, _film->audio_frame_rate ()));
-
-#ifdef DCPOMATIC_HAVE_PATCHED_FFMPEG
-       void* eb = _ebur128->get("Parsed_ebur128_0")->priv;
-       double true_peak = 0;
+       vector<AudioAnalysis::PeakTime> sample_peak;
        for (int i = 0; i < _film->audio_channels(); ++i) {
-               true_peak = max (true_peak, av_ebur128_get_true_peaks(eb)[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));
        }
-       _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 = dynamic_pointer_cast<const AudioContent> (_playlist->content().front ());
+               shared_ptr<const AudioContent> ac = _playlist->content().front()->audio;
                DCPOMATIC_ASSERT (ac);
-               _analysis->set_analysis_gain (ac->audio_gain ());
+               _analysis->set_analysis_gain (ac->gain ());
        }
 
        _analysis->write (_film->audio_analysis_path (_playlist));
@@ -152,8 +163,14 @@ AnalyseAudioJob::run ()
 }
 
 void
-AnalyseAudioJob::analyse (shared_ptr<const AudioBuffers> b)
+AnalyseAudioJob::analyse (shared_ptr<const AudioBuffers> b, DCPTime time)
 {
+#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 ();
 
@@ -163,16 +180,16 @@ AnalyseAudioJob::analyse (shared_ptr<const AudioBuffers> b)
                        float s = data[i];
                        float as = fabsf (s);
                        if (as < 10e-7) {
-                               /* SafeStringStream can't serialise and recover inf or -inf, so prevent such
+                               /* 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) {
-                               _sample_peak = as;
-                               _sample_peak_frame = _done + i;
+                       if (as > _sample_peak[j]) {
+                               _sample_peak[j] = as;
+                               _sample_peak_frame[j] = _done + i;
                        }
 
                        if (((_done + i) % _samples_per_point) == 0) {
@@ -184,4 +201,8 @@ AnalyseAudioJob::analyse (shared_ptr<const AudioBuffers> b)
        }
 
        _done += frames;
+
+       DCPTime const start = _playlist->start().get_value_or (DCPTime ());
+       DCPTime const length = _playlist->length ();
+       set_progress ((time.seconds() - start.seconds()) / (length.seconds() - start.seconds()));
 }