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-rw-r--r--src/lib/audio_analyser.cc92
-rw-r--r--src/lib/audio_analyser.h12
2 files changed, 52 insertions, 52 deletions
diff --git a/src/lib/audio_analyser.cc b/src/lib/audio_analyser.cc
index 7d4ee6ace..9597bbb14 100644
--- a/src/lib/audio_analyser.cc
+++ b/src/lib/audio_analyser.cc
@@ -53,15 +53,15 @@ static auto constexpr num_points = 1024;
AudioAnalyser::AudioAnalyser(shared_ptr<const Film> film, shared_ptr<const Playlist> playlist, bool whole_film, std::function<void (float)> set_progress)
- : _film (film)
- , _playlist (playlist)
- , _set_progress (set_progress)
+ : _film(film)
+ , _playlist(playlist)
+ , _set_progress(set_progress)
#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
, _ebur128(film->audio_frame_rate(), film->audio_channels())
#endif
- , _sample_peak (film->audio_channels())
- , _sample_peak_frame (film->audio_channels())
- , _analysis (film->audio_channels())
+ , _sample_peak(film->audio_channels())
+ , _sample_peak_frame(film->audio_channels())
+ , _analysis(film->audio_channels())
{
#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
@@ -102,18 +102,18 @@ AudioAnalyser::AudioAnalyser(shared_ptr<const Film> film, shared_ptr<const Playl
/* XXX: is this right? Especially for more than 5.1? */
vector<double> channel_corrections(_leqm_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
+ 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(
_leqm_channels,
@@ -125,18 +125,18 @@ AudioAnalyser::AudioAnalyser(shared_ptr<const Film> film, shared_ptr<const Playl
boost::thread::hardware_concurrency()
));
- DCPTime const length = _playlist->length (_film);
+ 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);
+ Frame const len = DCPTime(length - _start).frames_round(film->audio_frame_rate());
+ _samples_per_point = max(int64_t(1), len / num_points);
}
void
-AudioAnalyser::analyse (shared_ptr<AudioBuffers> b, DCPTime time)
+AudioAnalyser::analyse(shared_ptr<AudioBuffers> b, DCPTime time)
{
LOG_DEBUG_AUDIO_ANALYSIS("AudioAnalyser received {} frames at {}", b->frames(), to_string(time));
- DCPOMATIC_ASSERT (time >= _start);
+ DCPOMATIC_ASSERT(time >= _start);
/* In bug #2364 we had a lot of frames arriving here (~47s worth) which
* caused an OOM error on Windows. Check for the number of frames being
* reasonable here to make sure we catch this if it happens again.
@@ -144,12 +144,12 @@ AudioAnalyser::analyse (shared_ptr<AudioBuffers> b, DCPTime time)
DCPOMATIC_ASSERT(b->frames() < 480000);
#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
- if (Config::instance()->analyse_ebur128 ()) {
+ if (Config::instance()->analyse_ebur128()) {
_ebur128.process(b);
}
#endif
- int const frames = b->frames ();
+ int const frames = b->frames();
vector<double> interleaved(frames * _leqm_channels);
for (int j = 0; j < _leqm_channels; ++j) {
@@ -159,14 +159,14 @@ AudioAnalyser::analyse (shared_ptr<AudioBuffers> b, DCPTime time)
interleaved[i * _leqm_channels + j] = s;
- float as = fabsf (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);
+ _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;
@@ -174,9 +174,9 @@ AudioAnalyser::analyse (shared_ptr<AudioBuffers> b, DCPTime time)
}
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 ();
+ _current[j][AudioPoint::RMS] = sqrt(_current[j][AudioPoint::RMS] / _samples_per_point);
+ _analysis.add_point(j, _current[j]);
+ _current[j] = AudioPoint();
}
}
}
@@ -185,33 +185,33 @@ AudioAnalyser::analyse (shared_ptr<AudioBuffers> b, DCPTime time)
_done += frames;
- DCPTime const length = _playlist->length (_film);
- _set_progress ((time.seconds() - _start.seconds()) / (length.seconds() - _start.seconds()));
+ 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 ()
+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 ()))
+ 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);
+ _analysis.set_sample_peak(sample_peak);
#ifdef DCPOMATIC_HAVE_EBUR128_PATCHED_FFMPEG
- if (Config::instance()->analyse_ebur128 ()) {
+ 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]);
+ 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));
+ _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
@@ -220,11 +220,11 @@ AudioAnalyser::finish ()
gain was when we analysed it.
*/
if (auto ac = _playlist->content().front()->audio) {
- _analysis.set_analysis_gain (ac->gain());
+ _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());
+ _analysis.set_samples_per_point(_samples_per_point);
+ _analysis.set_sample_rate(_film->audio_frame_rate());
+ _analysis.set_leqm(_leqm->leq_m());
}
diff --git a/src/lib/audio_analyser.h b/src/lib/audio_analyser.h
index 3d40f8026..72826669c 100644
--- a/src/lib/audio_analyser.h
+++ b/src/lib/audio_analyser.h
@@ -41,18 +41,18 @@ class AudioAnalyser
public:
AudioAnalyser(std::shared_ptr<const Film> film, std::shared_ptr<const Playlist> playlist, bool whole_film, std::function<void (float)> set_progress);
- AudioAnalyser (AudioAnalyser const&) = delete;
- AudioAnalyser& operator= (AudioAnalyser const&) = delete;
+ AudioAnalyser(AudioAnalyser const&) = delete;
+ AudioAnalyser& operator=(AudioAnalyser const&) = delete;
- void analyse (std::shared_ptr<AudioBuffers>, dcpomatic::DCPTime time);
+ void analyse(std::shared_ptr<AudioBuffers>, dcpomatic::DCPTime time);
- dcpomatic::DCPTime start () const {
+ dcpomatic::DCPTime start() const {
return _start;
}
- void finish ();
+ void finish();
- AudioAnalysis get () const {
+ AudioAnalysis get() const {
return _analysis;
}
generated by cgit v1.2.3 (git 2.46.0) at 2026-04-05 20:39:24 +0000