X-Git-Url: https://main.carlh.net/gitweb/?p=dcpomatic.git;a=blobdiff_plain;f=src%2Flib%2Fanalyse_audio_job.cc;h=3e771d3f6af6a35c1e7a9e6a45b2be48b1b6b1ea;hp=af58e77ac59058097be12614c8c8499656ccff93;hb=422be0eece2bf6ee80db1d3c21553cd82efff789;hpb=94201bd2a5a4cb391b7f2bdeba56b928fed7cfe1 diff --git a/src/lib/analyse_audio_job.cc b/src/lib/analyse_audio_job.cc index af58e77ac..3e771d3f6 100644 --- a/src/lib/analyse_audio_job.cc +++ b/src/lib/analyse_audio_job.cc @@ -1,28 +1,42 @@ /* - Copyright (C) 2012 Carl Hetherington + Copyright (C) 2012-2015 Carl Hetherington - 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 . */ #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 +#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG +#include +#endif +} +#include +#include #include "i18n.h" @@ -31,16 +45,34 @@ using std::max; 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 f, shared_ptr c) - : Job (f) - , _content (c) +AnalyseAudioJob::AnalyseAudioJob (shared_ptr film, shared_ptr playlist) + : Job (film) + , _playlist (playlist) , _done (0) , _samples_per_point (1) + , _current (0) + , _sample_peak (0) + , _sample_peak_frame (0) +#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG + , _ebur128 (new AudioFilterGraph (film->audio_frame_rate(), film->audio_channels())) +#endif { +#ifdef DCPOMATIC_HAVE_EBUR128_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 (i); + } + delete[] _current; } string @@ -58,30 +90,69 @@ AnalyseAudioJob::json_name () const void AnalyseAudioJob::run () { - shared_ptr content = _content.lock (); - if (!content) { - return; - } + shared_ptr player (new Player (_film, _playlist)); + player->set_ignore_video (); + player->set_fast (); + player->set_play_referenced (); - shared_ptr playlist (new Playlist); - playlist->add (content); - shared_ptr player (new Player (_film, playlist)); - - int64_t const len = _film->length().frames (_film->audio_frame_rate()); + 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); - _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 c, _playlist->content ()) { + if (c->audio) { + 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 audio = player->get_audio (t, block, false); +#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG + if (Config::instance()->analyse_ebur128 ()) { + _ebur128->process (audio); + } +#endif + 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 ())); + +#ifdef DCPOMATIC_HAVE_EBUR128_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 ac = _playlist->content().front()->audio; + DCPOMATIC_ASSERT (ac); + _analysis->set_analysis_gain (ac->gain ()); } - _analysis->write (content->audio_analysis_path ()); - + _analysis->write (_film->audio_analysis_path (_playlist)); + set_progress (1); set_state (FINISHED_OK); } @@ -89,26 +160,34 @@ AnalyseAudioJob::run () void AnalyseAudioJob::analyse (shared_ptr 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) { + /* locked_stringstream 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; +}