Split audio analysis code off from the job.

1 files changed, 221 insertions, 0 deletions

diff --git a/src/lib/audio_analyser.cc b/src/lib/audio_analyser.cc
new file mode 100644
index 000000000..3caa997df
--- /dev/null
+++ b/src/lib/audio_analyser.cc
@@ -0,0 +1,221 @@
+/*
+    Copyright (C) 2021 Carl Hetherington <cth@carlh.net>
+
+    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.
+
+    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 DCP-o-matic.  If not, see <http://www.gnu.org/licenses/>.
+
+*/
+
+
+#include "audio_analyser.h"
+#include "audio_analysis.h"
+#include "audio_buffers.h"
+#include "audio_content.h"
+#include "audio_filter_graph.h"
+#include "audio_point.h"
+#include "config.h"
+#include "dcpomatic_log.h"
+#include "film.h"
+#include "filter.h"
+#include "playlist.h"
+#include "types.h"
+extern "C" {
+#include <leqm_nrt.h>
+#include <libavutil/channel_layout.h>
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+#include <libavfilter/f_ebur128.h>
+#endif
+}
+
+
+using std::make_shared;
+using std::max;
+using std::shared_ptr;
+using std::vector;
+using namespace dcpomatic;
+
+
+static auto constexpr num_points = 1024;
+
+
+AudioAnalyser::AudioAnalyser (shared_ptr<const Film> film, shared_ptr<const Playlist> playlist, bool from_zero, std::function<void (float)> set_progress)
+	: _film (film)
+	, _playlist (playlist)
+	, _set_progress (set_progress)
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+	, _ebur128 (new AudioFilterGraph(film->audio_frame_rate(), film->audio_channels()))
+#endif
+	, _sample_peak (new float[film->audio_channels()])
+	, _sample_peak_frame (new Frame[film->audio_channels()])
+	, _analysis (film->audio_channels())
+{
+
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+	_filters.push_back (new Filter("ebur128", "ebur128", "audio", "ebur128=peak=true"));
+	_ebur128->setup (_filters);
+#endif
+
+	_current = new AudioPoint[_film->audio_channels()];
+
+	if (!from_zero) {
+		_start = _playlist->start().get_value_or(DCPTime());
+	}
+
+	for (int i = 0; i < film->audio_channels(); ++i) {
+		_sample_peak[i] = 0;
+		_sample_peak_frame[i] = 0;
+	}
+
+	auto add_if_required = [](vector<double>& v, size_t i, double db) {
+		if (v.size() > i) {
+			v[i] = pow(10, db / 20);
+		}
+	};
+
+	/* 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()
+		));
+
+	DCPTime const length = _playlist->length (_film);
+
+	Frame const len = DCPTime (length - _start).frames_round (film->audio_frame_rate());
+	_samples_per_point = max (int64_t (1), len / num_points);
+}
+
+
+AudioAnalyser::~AudioAnalyser ()
+{
+	delete[] _current;
+	for (auto i: _filters) {
+		delete const_cast<Filter*> (i);
+	}
+	delete[] _sample_peak;
+	delete[] _sample_peak_frame;
+}
+
+
+void
+AudioAnalyser::analyse (shared_ptr<const AudioBuffers> b, DCPTime time)
+{
+	LOG_DEBUG_AUDIO_ANALYSIS("Received %1 frames at %2", b->frames(), to_string(time));
+	DCPOMATIC_ASSERT (time >= _start);
+
+#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
+	if (Config::instance()->analyse_ebur128 ()) {
+		_ebur128->process (b);
+	}
+#endif
+
+	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
+				   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[j]) {
+				_sample_peak[j] = as;
+				_sample_peak_frame[j] = _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 ();
+			}
+		}
+	}
+
+	_leqm->add(interleaved);
+
+	_done += frames;
+
+	DCPTime const length = _playlist->length (_film);
+	_set_progress ((time.seconds() - _start.seconds()) / (length.seconds() - _start.seconds()));
+	LOG_DEBUG_AUDIO_ANALYSIS_NC("Frames processed");
+}
+
+
+void
+AudioAnalyser::finish ()
+{
+	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_EBUR128_PATCHED_FFMPEG
+	if (Config::instance()->analyse_ebur128 ()) {
+		void* eb = _ebur128->get("Parsed_ebur128_0")->priv;
+		vector<float> true_peak;
+		for (int i = 0; i < _film->audio_channels(); ++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));
+		_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.
+		*/
+		if (auto ac = _playlist->content().front()->audio) {
+			_analysis.set_analysis_gain (ac->gain());
+		}
+	}
+
+	_analysis.set_samples_per_point (_samples_per_point);
+	_analysis.set_sample_rate (_film->audio_frame_rate ());
+	_analysis.set_leqm (_leqm->leq_m());
+}