Revert mac rdisk vs disk stuff as I'm not convinced it makes much difference.

[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 1378b66a433fc0fa5a531d4db682f4f26ea3de31..e77f83bdedfd8c8b82dd21f8c28d287f2b8b8ab8 100644 (file)
--- a/src/lib/analyse_audio_job.cc
+++ b/src/lib/analyse_audio_job.cc
@@ -1,5 +1,5 @@
 /*
-    Copyright (C) 2012-2015 Carl Hetherington <cth@carlh.net>
+    Copyright (C) 2012-2018 Carl Hetherington <cth@carlh.net>
 
     This file is part of DCP-o-matic.
 
@@ -30,6 +30,7 @@
 #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>
@@ -47,12 +48,23 @@ using std::min;
 using std::cout;
 using boost::shared_ptr;
 using boost::dynamic_pointer_cast;
+using namespace dcpomatic;
 
 int const AnalyseAudioJob::_num_points = 1024;
 
-AnalyseAudioJob::AnalyseAudioJob (shared_ptr<const Film> film, shared_ptr<const Playlist> playlist)
+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)
@@ -71,10 +83,40 @@ AnalyseAudioJob::AnalyseAudioJob (shared_ptr<const Film> film, shared_ptr<const
                _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);
        }
@@ -86,7 +128,7 @@ AnalyseAudioJob::~AnalyseAudioJob ()
 string
 AnalyseAudioJob::name () const
 {
-       return _("Analyse audio");
+       return _("Analysing audio");
 }
 
 string
@@ -98,16 +140,16 @@ AnalyseAudioJob::json_name () const
 void
 AnalyseAudioJob::run ()
 {
-       shared_ptr<Player> player (new Player (_film, _playlist));
+       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 start = _playlist->start().get_value_or (DCPTime ());
-       DCPTime const length = _playlist->length ();
+       DCPTime const length = _playlist->length (_film);
 
-       Frame const len = DCPTime (length - start).frames_round (_film->audio_frame_rate());
+       Frame const len = DCPTime (length - _start).frames_round (_film->audio_frame_rate());
        _samples_per_point = max (int64_t (1), len / _num_points);
 
        delete[] _current;
@@ -122,6 +164,7 @@ AnalyseAudioJob::run ()
        }
 
        if (has_any_audio) {
+               player->seek (_start, true);
                _done = 0;
                while (!player->pass ()) {}
        }
@@ -152,11 +195,15 @@ AnalyseAudioJob::run ()
                   gain was when we analysed it.
                */
                shared_ptr<const AudioContent> ac = _playlist->content().front()->audio;
-               DCPOMATIC_ASSERT (ac);
-               _analysis->set_analysis_gain (ac->gain ());
+               if (ac) {
+                       _analysis->set_analysis_gain (ac->gain());
+               }
        }
 
-       _analysis->write (_film->audio_analysis_path (_playlist));
+       _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);
 
        set_progress (1);
        set_state (FINISHED_OK);
@@ -165,6 +212,8 @@ AnalyseAudioJob::run ()
 void
 AnalyseAudioJob::analyse (shared_ptr<const AudioBuffers> b, DCPTime time)
 {
+       DCPOMATIC_ASSERT (time >= _start);
+
 #ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
        if (Config::instance()->analyse_ebur128 ()) {
                _ebur128->process (b);
@@ -173,11 +222,15 @@ AnalyseAudioJob::analyse (shared_ptr<const AudioBuffers> b, DCPTime time)
 
        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
@@ -200,9 +253,10 @@ AnalyseAudioJob::analyse (shared_ptr<const AudioBuffers> b, DCPTime time)
                }
        }
 
+       _leqm->add(interleaved);
+
        _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()));
+       DCPTime const length = _playlist->length (_film);
+       set_progress ((time.seconds() - _start.seconds()) / (length.seconds() - _start.seconds()));
 }