#include "analyse_audio_job.h"
#include "audio_content.h"
#include "compose.hpp"
+#include "dcpomatic_log.h"
#include "film.h"
#include "player.h"
#include "playlist.h"
#include "audio_filter_graph.h"
#include "config.h"
extern "C" {
+#include <leqm_nrt.h>
#include <libavutil/channel_layout.h>
#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
#include <libavfilter/f_ebur128.h>
#endif
}
-#include <boost/foreach.hpp>
#include <iostream>
#include "i18n.h"
using std::max;
using std::min;
using std::cout;
-using boost::shared_ptr;
-using boost::dynamic_pointer_cast;
+using std::shared_ptr;
+using std::dynamic_pointer_cast;
+using namespace dcpomatic;
+#if BOOST_VERSION >= 106100
+using namespace boost::placeholders;
+#endif
int const AnalyseAudioJob::_num_points = 1024;
+static void add_if_required(vector<double>& v, size_t i, double db)
+{
+ if (v.size() > i) {
+ v[i] = pow(10, db / 20);
+ }
+}
+
/** @param from_zero true to analyse audio from time 0 in the playlist, otherwise begin at Playlist::start */
AnalyseAudioJob::AnalyseAudioJob (shared_ptr<const Film> film, shared_ptr<const Playlist> playlist, bool from_zero)
: Job (film)
, _playlist (playlist)
+ , _path (film->audio_analysis_path(playlist))
, _from_zero (from_zero)
, _done (0)
, _samples_per_point (1)
, _ebur128 (new AudioFilterGraph (film->audio_frame_rate(), film->audio_channels()))
#endif
{
+ LOG_DEBUG_AUDIO_ANALYSIS_NC("AnalyseAudioJob::AnalyseAudioJob");
+
#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
_filters.push_back (new Filter ("ebur128", "ebur128", "audio", "ebur128=peak=true"));
_ebur128->setup (_filters);
if (!_from_zero) {
_start = _playlist->start().get_value_or(DCPTime());
}
+
+ /* 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()
+ ));
}
AnalyseAudioJob::~AnalyseAudioJob ()
{
- BOOST_FOREACH (Filter const * i, _filters) {
+ stop_thread ();
+ for (auto i: _filters) {
delete const_cast<Filter*> (i);
}
delete[] _current;
string
AnalyseAudioJob::name () const
{
- return _("Analyse audio");
+ return _("Analysing audio");
}
string
void
AnalyseAudioJob::run ()
{
- shared_ptr<Player> player (new Player (_film, _playlist));
+ LOG_DEBUG_AUDIO_ANALYSIS_NC("AnalyseAudioJob::run");
+
+ shared_ptr<Player> player (new Player(_film, _playlist));
player->set_ignore_video ();
- player->set_ignore_subtitle ();
+ player->set_ignore_text ();
player->set_fast ();
player->set_play_referenced ();
player->Audio.connect (bind (&AnalyseAudioJob::analyse, this, _1, _2));
- DCPTime const length = _playlist->length ();
+ 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);
_analysis.reset (new AudioAnalysis (_film->audio_channels ()));
bool has_any_audio = false;
- BOOST_FOREACH (shared_ptr<Content> c, _playlist->content ()) {
+ for (auto c: _playlist->content()) {
if (c->audio) {
has_any_audio = true;
}
}
if (has_any_audio) {
+ LOG_DEBUG_AUDIO_ANALYSIS("Seeking to %1", to_string(_start));
player->seek (_start, true);
_done = 0;
+ LOG_DEBUG_AUDIO_ANALYSIS("Starting loop for playlist of length %1", to_string(length));
while (!player->pass ()) {}
}
+ LOG_DEBUG_AUDIO_ANALYSIS_NC("Loop complete");
+
vector<AudioAnalysis::PeakTime> sample_peak;
for (int i = 0; i < _film->audio_channels(); ++i) {
sample_peak.push_back (
gain was when we analysed it.
*/
shared_ptr<const AudioContent> ac = _playlist->content().front()->audio;
- DCPOMATIC_ASSERT (ac);
- _analysis->set_analysis_gain (ac->gain ());
+ if (ac) {
+ _analysis->set_analysis_gain (ac->gain());
+ }
}
_analysis->set_samples_per_point (_samples_per_point);
_analysis->set_sample_rate (_film->audio_frame_rate ());
- _analysis->write (_film->audio_analysis_path (_playlist));
+ _analysis->set_leqm (_leqm->leq_m());
+ _analysis->write (_path);
+ LOG_DEBUG_AUDIO_ANALYSIS_NC("Job finished");
set_progress (1);
set_state (FINISHED_OK);
}
void
AnalyseAudioJob::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
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
}
}
+ _leqm->add(interleaved);
+
_done += frames;
- DCPTime const length = _playlist->length ();
+ DCPTime const length = _playlist->length (_film);
set_progress ((time.seconds() - _start.seconds()) / (length.seconds() - _start.seconds()));
+ LOG_DEBUG_AUDIO_ANALYSIS_NC("Frames processed");
}
projects / dcpomatic.git / blobdiff