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/*
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
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,
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.
*/
#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"
extern "C" {
#include <libavutil/channel_layout.h>
}
#include <boost/foreach.hpp>
#include <iostream>
#include "i18n.h"
using std::string;
using std::max;
using std::min;
using std::cout;
using boost::shared_ptr;
using boost::dynamic_pointer_cast;
int const AnalyseAudioJob::_num_points = 1024;
extern "C" {
/* I added these functions to the FFmpeg that DCP-o-matic is built with, but there isn't an
existing header file to put the prototype in. Cheat by putting it in here.
*/
double* av_ebur128_get_true_peaks (void* context);
double* av_ebur128_get_sample_peaks (void* context);
double av_ebur128_get_integrated_loudness(void* context);
double av_ebur128_get_loudness_range (void* context);
}
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)
, _ebur128 (new AudioFilterGraph (film->audio_frame_rate(), av_get_default_channel_layout(film->audio_channels())))
{
_filters.push_back (new Filter ("ebur128", "ebur128", "audio", "ebur128=peak=true"));
_ebur128->setup (_filters);
}
AnalyseAudioJob::~AnalyseAudioJob ()
{
BOOST_FOREACH (Filter const * i, _filters) {
delete const_cast<Filter*> (i);
}
delete[] _current;
}
string
AnalyseAudioJob::name () const
{
return _("Analyse audio");
}
string
AnalyseAudioJob::json_name () const
{
return N_("analyse_audio");
}
void
AnalyseAudioJob::run ()
{
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 ();
Frame const len = DCPTime (length - start).frames_round (_film->audio_frame_rate());
_samples_per_point = max (int64_t (1), len / _num_points);
delete[] _current;
_current = new AudioPoint[_film->audio_channels ()];
_analysis.reset (new AudioAnalysis (_film->audio_channels ()));
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);
_ebur128->process (audio);
analyse (audio);
set_progress ((t.seconds() - start.seconds()) / (length.seconds() - start.seconds()));
}
}
_analysis->set_sample_peak (_sample_peak, DCPTime::from_frames (_sample_peak_frame, _film->audio_frame_rate ()));
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));
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)
{
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 = 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 (((_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 += frames;
}
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