X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Fanalyse_audio_job.cc;h=769f3762bee98ee214c9972ddb0c6018c0cc8767;hb=be8675cc978fbd52817943e88ea40b81f8a4c9c4;hp=2146b03c58aa2dbd7a7fa3c825bfd29244382bb7;hpb=c5b9ad09ab5eaf032b0816f619ab5d75254e8597;p=dcpomatic.git

diff --git a/src/lib/analyse_audio_job.cc b/src/lib/analyse_audio_job.cc
index 2146b03c5..769f3762b 100644
--- a/src/lib/analyse_audio_job.cc
+++ b/src/lib/analyse_audio_job.cc
@@ -20,11 +20,21 @@
 #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>
+#ifdef DCPOMATIC_HAVE_PATCHED_FFMPEG
+#include <libavfilter/f_ebur128.h>
+#endif
+}
 #include <boost/foreach.hpp>
+#include <iostream>
 
 #include "i18n.h"
 
@@ -42,10 +52,25 @@ AnalyseAudioJob::AnalyseAudioJob (shared_ptr<const Film> film, shared_ptr<const
 	, _playlist (playlist)
 	, _done (0)
 	, _samples_per_point (1)
-	, _overall_peak (0)
-	, _overall_peak_frame (0)
+	, _current (0)
+	, _sample_peak (0)
+	, _sample_peak_frame (0)
+#ifdef DCPOMATIC_HAVE_PATCHED_FFMPEG
+	, _ebur128 (new AudioFilterGraph (film->audio_frame_rate(), av_get_default_channel_layout(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
@@ -65,11 +90,17 @@ AnalyseAudioJob::run ()
 {
 	shared_ptr<Player> player (new Player (_film, _playlist));
 	player->set_ignore_video ();
+	player->set_fast ();
+	player->set_play_referenced ();
 
-	int64_t const len = _playlist->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 ()));
 
 	bool has_any_audio = false;
@@ -82,14 +113,39 @@ AnalyseAudioJob::run ()
 	if (has_any_audio) {
 		_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());
+		for (DCPTime t = start; t < length; t += block) {
+			shared_ptr<const AudioBuffers> audio = player->get_audio (t, block, false);
+#ifdef DCPOMATIC_HAVE_PATCHED_FFMPEG
+			_ebur128->process (audio);
+#endif
+			analyse (audio);
+			set_progress ((t.seconds() - start.seconds()) / (length.seconds() - start.seconds()));
 		}
 	}
 
-	_analysis->set_peak (_overall_peak, DCPTime::from_frames (_overall_peak_frame, _film->audio_frame_rate ()));
-	_analysis->write (_film->audio_analysis_path ());
+	_analysis->set_sample_peak (_sample_peak, DCPTime::from_frames (_sample_peak_frame, _film->audio_frame_rate ()));
+
+#ifdef DCPOMATIC_HAVE_PATCHED_FFMPEG
+	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);
@@ -98,34 +154,34 @@ AnalyseAudioJob::run ()
 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) {
+	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));
-
-			float const as = fabs (s);
-
 			_current[j][AudioPoint::PEAK] = max (_current[j][AudioPoint::PEAK], as);
 
-			if (as > _overall_peak) {
-				_overall_peak = as;
-				_overall_peak_frame = _done + i;
+			if (as > _sample_peak) {
+				_sample_peak = as;
+				_sample_peak_frame = _done + i;
 			}
 
-			if ((_done % _samples_per_point) == 0) {
+			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;
 }