/*
- Copyright (C) 2012 Carl Hetherington <cth@carlh.net>
+ Copyright (C) 2012-2015 Carl Hetherington <cth@carlh.net>
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
#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_PATCHED_FFMPEG
+#include <libavfilter/f_ebur128.h>
+#endif
+}
+#include <boost/foreach.hpp>
+#include <iostream>
#include "i18n.h"
using std::min;
using std::cout;
using boost::shared_ptr;
+using boost::dynamic_pointer_cast;
int const AnalyseAudioJob::_num_points = 1024;
-AnalyseAudioJob::AnalyseAudioJob (shared_ptr<const Film> f, shared_ptr<AudioContent> c)
- : Job (f)
- , _content (c)
+AnalyseAudioJob::AnalyseAudioJob (shared_ptr<const Film> film, shared_ptr<const Playlist> playlist)
+ : Job (film)
+ , _playlist (playlist)
, _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(), film->audio_channels()))
+#endif
{
+#ifdef DCPOMATIC_HAVE_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
void
AnalyseAudioJob::run ()
{
- shared_ptr<AudioContent> content = _content.lock ();
- if (!content) {
- return;
- }
+ shared_ptr<Player> player (new Player (_film, _playlist));
+ player->set_ignore_video ();
+ player->set_fast ();
+ player->set_play_referenced ();
+
+ DCPTime const start = _playlist->start().get_value_or (DCPTime ());
+ DCPTime const length = _playlist->length ();
- shared_ptr<Playlist> playlist (new Playlist);
- playlist->add (content);
- shared_ptr<Player> player (new Player (_film, playlist));
-
- int64_t const len = _film->length().frames (_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);
- _current.resize (_film->audio_channels ());
+ delete[] _current;
+ _current = new AudioPoint[_film->audio_channels ()];
_analysis.reset (new AudioAnalysis (_film->audio_channels ()));
- _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());
+ bool has_any_audio = false;
+ BOOST_FOREACH (shared_ptr<Content> c, _playlist->content ()) {
+ if (dynamic_pointer_cast<AudioContent> (c)) {
+ 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
+ if (Config::instance()->analyse_ebur128 ()) {
+ _ebur128->process (audio);
+ }
+#endif
+ analyse (audio);
+ set_progress ((t.seconds() - start.seconds()) / (length.seconds() - start.seconds()));
+ }
}
- _analysis->write (content->audio_analysis_path ());
-
+ _analysis->set_sample_peak (_sample_peak, DCPTime::from_frames (_sample_peak_frame, _film->audio_frame_rate ()));
+
+#ifdef DCPOMATIC_HAVE_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 = dynamic_pointer_cast<const AudioContent> (_playlist->content().front ());
+ DCPOMATIC_ASSERT (ac);
+ _analysis->set_analysis_gain (ac->audio_gain ());
+ }
+
+ _analysis->write (_film->audio_analysis_path (_playlist));
+
set_progress (1);
set_state (FINISHED_OK);
}
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) {
- /* stringstream 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) {
+ /* SafeStringStream 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));
+ _current[j][AudioPoint::PEAK] = max (_current[j][AudioPoint::PEAK], as);
- if ((_done % _samples_per_point) == 0) {
+ if (as > _sample_peak) {
+ _sample_peak = as;
+ _sample_peak_frame = _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 ();
}
}
-
- ++_done;
}
-}
+ _done += frames;
+}