/*
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 "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
}
#include "i18n.h"
using std::string;
+using std::vector;
using std::max;
using std::min;
using std::cout;
, _done (0)
, _samples_per_point (1)
, _current (0)
- , _sample_peak (0)
- , _sample_peak_frame (0)
-#ifdef DCPOMATIC_HAVE_PATCHED_FFMPEG
+ , _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 ()
delete const_cast<Filter*> (i);
}
delete[] _current;
+ delete[] _sample_peak;
+ delete[] _sample_peak_frame;
}
string
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;
}
}
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
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
if (Config::instance()->analyse_ebur128 ()) {
_ebur128->process (audio);
}
}
}
- _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_PATCHED_FFMPEG
+#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));
/* 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));
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) {