Add channel details to high-audio-level hints (#822).

[dcpomatic.git] / src / lib / analyse_audio_job.cc

diff --git a/src/lib/analyse_audio_job.cc b/src/lib/analyse_audio_job.cc
index 9a24a9188bfc708bc74e92d27f16f70f2acf7d32..9fce354df81d13054c5bc1efea2240c0db267a5a 100644 (file)
--- a/src/lib/analyse_audio_job.cc
+++ b/src/lib/analyse_audio_job.cc
@@ -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/>.
 
 */
 
@@ -40,6 +41,7 @@ extern "C" {
 #include "i18n.h"
 
 using std::string;
+using std::vector;
 using std::max;
 using std::min;
 using std::cout;
@@ -54,8 +56,8 @@ 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)
+       , _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
@@ -72,6 +74,8 @@ AnalyseAudioJob::~AnalyseAudioJob ()
                delete const_cast<Filter*> (i);
        }
        delete[] _current;
+       delete[] _sample_peak;
+       delete[] _sample_peak_frame;
 }
 
 string
@@ -126,14 +130,20 @@ AnalyseAudioJob::run ()
                }
        }
 
-       _analysis->set_sample_peak (_sample_peak, DCPTime::from_frames (_sample_peak_frame, _film->audio_frame_rate ()));
+       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;
-               double true_peak = 0;
+               vector<float> true_peak;
                for (int i = 0; i < _film->audio_channels(); ++i) {
-                       true_peak = max (true_peak, av_ebur128_get_true_peaks(eb)[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));
@@ -147,7 +157,7 @@ AnalyseAudioJob::run ()
                */
                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));
@@ -168,16 +178,16 @@ 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) {