Paul Davis
b35518e212
This is mostly a simple lexical search+replace but the absence of operator< for std::weak_ptr<T> leads to some complications, particularly with Evoral::Sequence and ExportPortChannel.
2116 lines
61 KiB
C++
2116 lines
61 KiB
C++
/*
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* Copyright (C) 2006-2014 David Robillard <d@drobilla.net>
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* Copyright (C) 2006-2017 Paul Davis <paul@linuxaudiosystems.com>
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* Copyright (C) 2007-2012 Carl Hetherington <carl@carlh.net>
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* Copyright (C) 2012-2019 Robin Gareus <robin@gareus.org>
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* Copyright (C) 2015-2018 Ben Loftis <ben@harrisonconsoles.com>
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* Copyright (C) 2016-2017 Nick Mainsbridge <mainsbridge@gmail.com>
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* Copyright (C) 2016 Tim Mayberry <mojofunk@gmail.com>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include <algorithm>
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#include <cfloat>
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#include <climits>
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#include <cmath>
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#include <memory>
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#include <set>
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#include <boost/scoped_array.hpp>
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#include <glibmm/fileutils.h>
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#include <glibmm/threads.h>
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#include "pbd/gstdio_compat.h"
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#include "pbd/basename.h"
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#include "pbd/xml++.h"
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#include "pbd/enumwriter.h"
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#include "pbd/convert.h"
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#include "evoral/Curve.h"
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#include "ardour/audioengine.h"
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#include "ardour/analysis_graph.h"
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#include "ardour/audioregion.h"
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#include "ardour/session.h"
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#include "ardour/dB.h"
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#include "ardour/debug.h"
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#include "ardour/event_type_map.h"
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#include "ardour/playlist.h"
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#include "ardour/audiofilesource.h"
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#include "ardour/region_factory.h"
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#include "ardour/runtime_functions.h"
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#include "ardour/sndfilesource.h"
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#include "ardour/transient_detector.h"
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#include "ardour/parameter_descriptor.h"
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#include "ardour/progress.h"
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#include "audiographer/general/interleaver.h"
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#include "audiographer/general/sample_format_converter.h"
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#include "audiographer/sndfile/sndfile_writer.h"
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#include "pbd/i18n.h"
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#include <locale.h>
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using namespace std;
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using namespace ARDOUR;
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using namespace PBD;
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#define S2SC(s) Temporal::samples_to_superclock (s, TEMPORAL_SAMPLE_RATE)
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#define SC2S(s) Temporal::superclock_to_samples (s, TEMPORAL_SAMPLE_RATE)
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namespace ARDOUR {
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namespace Properties {
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PBD::PropertyDescriptor<bool> envelope_active;
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PBD::PropertyDescriptor<bool> default_fade_in;
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PBD::PropertyDescriptor<bool> default_fade_out;
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PBD::PropertyDescriptor<bool> fade_in_active;
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PBD::PropertyDescriptor<bool> fade_out_active;
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PBD::PropertyDescriptor<float> scale_amplitude;
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PBD::PropertyDescriptor<std::shared_ptr<AutomationList> > fade_in;
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PBD::PropertyDescriptor<std::shared_ptr<AutomationList> > inverse_fade_in;
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PBD::PropertyDescriptor<std::shared_ptr<AutomationList> > fade_out;
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PBD::PropertyDescriptor<std::shared_ptr<AutomationList> > inverse_fade_out;
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PBD::PropertyDescriptor<std::shared_ptr<AutomationList> > envelope;
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}
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}
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/* Curve manipulations */
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static void
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reverse_curve (std::shared_ptr<Evoral::ControlList> dst, std::shared_ptr<const Evoral::ControlList> src)
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{
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const timepos_t end = src->when(false);
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// TODO read-lock of src (!)
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for (Evoral::ControlList::const_reverse_iterator it = src->rbegin(); it!=src->rend(); it++) {
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/* ugh ... the double "distance" calls (with totally different
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semantics ... horrible
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*/
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dst->fast_simple_add (timepos_t ((*it)->when.distance (end)), (*it)->value);
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}
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}
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static void
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generate_inverse_power_curve (std::shared_ptr<Evoral::ControlList> dst, std::shared_ptr<const Evoral::ControlList> src)
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{
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// calc inverse curve using sum of squares
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for (Evoral::ControlList::const_iterator it = src->begin(); it!=src->end(); ++it ) {
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float value = (*it)->value;
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value = 1 - powf(value,2);
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value = sqrtf(value);
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dst->fast_simple_add ((*it)->when, value );
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}
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}
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static void
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generate_db_fade (std::shared_ptr<Evoral::ControlList> dst, double len, int num_steps, float dB_drop)
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{
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dst->clear ();
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dst->fast_simple_add (timepos_t (Temporal::AudioTime), 1);
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//generate a fade-out curve by successively applying a gain drop
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float fade_speed = dB_to_coefficient(dB_drop / (float) num_steps);
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float coeff = GAIN_COEFF_UNITY;
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for (int i = 1; i < (num_steps-1); i++) {
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coeff *= fade_speed;
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dst->fast_simple_add (timepos_t (samplepos_t (len*(double)i/(double)num_steps)), coeff);
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}
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dst->fast_simple_add (timepos_t ((samplepos_t)len), GAIN_COEFF_SMALL);
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}
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static void
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merge_curves (std::shared_ptr<Evoral::ControlList> dst,
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std::shared_ptr<const Evoral::ControlList> curve1,
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std::shared_ptr<const Evoral::ControlList> curve2)
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{
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Evoral::ControlList::EventList::size_type size = curve1->size();
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//curve lengths must match for now
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if (size != curve2->size()) {
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return;
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}
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Evoral::ControlList::const_iterator c1 = curve1->begin();
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int count = 0;
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for (Evoral::ControlList::const_iterator c2 = curve2->begin(); c2!=curve2->end(); c2++ ) {
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float v1 = accurate_coefficient_to_dB((*c1)->value);
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float v2 = accurate_coefficient_to_dB((*c2)->value);
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double interp = v1 * ( 1.0-( (double)count / (double)size) );
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interp += v2 * ( (double)count / (double)size );
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interp = dB_to_coefficient(interp);
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dst->fast_simple_add ((*c1)->when, interp );
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c1++;
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count++;
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}
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}
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void
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AudioRegion::make_property_quarks ()
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{
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Properties::envelope_active.property_id = g_quark_from_static_string (X_("envelope-active"));
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DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for envelope-active = %1\n", Properties::envelope_active.property_id));
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Properties::default_fade_in.property_id = g_quark_from_static_string (X_("default-fade-in"));
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DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for default-fade-in = %1\n", Properties::default_fade_in.property_id));
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Properties::default_fade_out.property_id = g_quark_from_static_string (X_("default-fade-out"));
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DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for default-fade-out = %1\n", Properties::default_fade_out.property_id));
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Properties::fade_in_active.property_id = g_quark_from_static_string (X_("fade-in-active"));
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DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for fade-in-active = %1\n", Properties::fade_in_active.property_id));
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Properties::fade_out_active.property_id = g_quark_from_static_string (X_("fade-out-active"));
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DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for fade-out-active = %1\n", Properties::fade_out_active.property_id));
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Properties::scale_amplitude.property_id = g_quark_from_static_string (X_("scale-amplitude"));
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DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for scale-amplitude = %1\n", Properties::scale_amplitude.property_id));
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Properties::fade_in.property_id = g_quark_from_static_string (X_("FadeIn"));
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DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for FadeIn = %1\n", Properties::fade_in.property_id));
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Properties::inverse_fade_in.property_id = g_quark_from_static_string (X_("InverseFadeIn"));
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DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for InverseFadeIn = %1\n", Properties::inverse_fade_in.property_id));
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Properties::fade_out.property_id = g_quark_from_static_string (X_("FadeOut"));
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DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for FadeOut = %1\n", Properties::fade_out.property_id));
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Properties::inverse_fade_out.property_id = g_quark_from_static_string (X_("InverseFadeOut"));
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DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for InverseFadeOut = %1\n", Properties::inverse_fade_out.property_id));
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Properties::envelope.property_id = g_quark_from_static_string (X_("Envelope"));
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DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for Envelope = %1\n", Properties::envelope.property_id));
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}
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void
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AudioRegion::register_properties ()
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{
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/* no need to register parent class properties */
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add_property (_envelope_active);
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add_property (_default_fade_in);
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add_property (_default_fade_out);
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add_property (_fade_in_active);
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add_property (_fade_out_active);
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add_property (_scale_amplitude);
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add_property (_fade_in);
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add_property (_inverse_fade_in);
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add_property (_fade_out);
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add_property (_inverse_fade_out);
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add_property (_envelope);
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}
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#define AUDIOREGION_STATE_DEFAULT \
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_envelope_active (Properties::envelope_active, false) \
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, _default_fade_in (Properties::default_fade_in, true) \
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, _default_fade_out (Properties::default_fade_out, true) \
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, _fade_in_active (Properties::fade_in_active, true) \
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, _fade_out_active (Properties::fade_out_active, true) \
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, _scale_amplitude (Properties::scale_amplitude, 1.0) \
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, _fade_in (Properties::fade_in, std::shared_ptr<AutomationList> (new AutomationList (Evoral::Parameter (FadeInAutomation), Temporal::AudioTime))) \
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, _inverse_fade_in (Properties::inverse_fade_in, std::shared_ptr<AutomationList> (new AutomationList (Evoral::Parameter (FadeInAutomation), Temporal::AudioTime))) \
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, _fade_out (Properties::fade_out, std::shared_ptr<AutomationList> (new AutomationList (Evoral::Parameter (FadeOutAutomation), Temporal::AudioTime))) \
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, _inverse_fade_out (Properties::inverse_fade_out, std::shared_ptr<AutomationList> (new AutomationList (Evoral::Parameter (FadeOutAutomation), Temporal::AudioTime)))
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#define AUDIOREGION_COPY_STATE(other) \
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_envelope_active (Properties::envelope_active, other->_envelope_active) \
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, _default_fade_in (Properties::default_fade_in, other->_default_fade_in) \
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, _default_fade_out (Properties::default_fade_out, other->_default_fade_out) \
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, _fade_in_active (Properties::fade_in_active, other->_fade_in_active) \
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, _fade_out_active (Properties::fade_out_active, other->_fade_out_active) \
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, _scale_amplitude (Properties::scale_amplitude, other->_scale_amplitude) \
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, _fade_in (Properties::fade_in, std::shared_ptr<AutomationList> (new AutomationList (*other->_fade_in.val()))) \
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, _inverse_fade_in (Properties::fade_in, std::shared_ptr<AutomationList> (new AutomationList (*other->_inverse_fade_in.val()))) \
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, _fade_out (Properties::fade_in, std::shared_ptr<AutomationList> (new AutomationList (*other->_fade_out.val()))) \
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, _inverse_fade_out (Properties::fade_in, std::shared_ptr<AutomationList> (new AutomationList (*other->_inverse_fade_out.val())))
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/* a Session will reset these to its chosen defaults by calling AudioRegion::set_default_fade() */
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void
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AudioRegion::init ()
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{
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register_properties ();
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suspend_property_changes();
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set_default_fades ();
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set_default_envelope ();
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resume_property_changes();
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listen_to_my_curves ();
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connect_to_analysis_changed ();
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connect_to_header_position_offset_changed ();
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}
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/** Constructor for use by derived types only */
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AudioRegion::AudioRegion (Session& s, timepos_t const & start, timecnt_t const & len, std::string name)
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: Region (s, start, len, name, DataType::AUDIO)
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, AUDIOREGION_STATE_DEFAULT
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, _envelope (Properties::envelope, std::shared_ptr<AutomationList> (new AutomationList (Evoral::Parameter(EnvelopeAutomation), Temporal::AudioTime)))
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, _automatable (s, Temporal::AudioTime)
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, _fade_in_suspended (0)
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, _fade_out_suspended (0)
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{
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init ();
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assert (_sources.size() == _master_sources.size());
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}
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/** Basic AudioRegion constructor */
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AudioRegion::AudioRegion (const SourceList& srcs)
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: Region (srcs)
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, AUDIOREGION_STATE_DEFAULT
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, _envelope (Properties::envelope, std::shared_ptr<AutomationList> (new AutomationList (Evoral::Parameter(EnvelopeAutomation), Temporal::AudioTime)))
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, _automatable(srcs[0]->session(), Temporal::AudioTime)
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, _fade_in_suspended (0)
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, _fade_out_suspended (0)
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{
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init ();
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assert (_sources.size() == _master_sources.size());
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}
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AudioRegion::AudioRegion (std::shared_ptr<const AudioRegion> other)
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: Region (other)
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, AUDIOREGION_COPY_STATE (other)
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/* As far as I can see, the _envelope's times are relative to region position, and have nothing
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* to do with sources (and hence _start). So when we copy the envelope, we just use the supplied offset.
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*/
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, _envelope (Properties::envelope, std::shared_ptr<AutomationList> (new AutomationList (*other->_envelope.val(), timepos_t (Temporal::AudioTime), other->len_as_tpos ())))
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, _automatable (other->session(), Temporal::AudioTime)
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, _fade_in_suspended (0)
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, _fade_out_suspended (0)
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{
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/* don't use init here, because we got fade in/out from the other region
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*/
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register_properties ();
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listen_to_my_curves ();
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connect_to_analysis_changed ();
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connect_to_header_position_offset_changed ();
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assert(_type == DataType::AUDIO);
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assert (_sources.size() == _master_sources.size());
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}
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AudioRegion::AudioRegion (std::shared_ptr<const AudioRegion> other, timecnt_t const & offset)
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: Region (other, offset)
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, AUDIOREGION_COPY_STATE (other)
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/* As far as I can see, the _envelope's times are relative to region position, and have nothing
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to do with sources (and hence _start). So when we copy the envelope, we just use the supplied offset.
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*/
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, _envelope (Properties::envelope, std::shared_ptr<AutomationList> (new AutomationList (*other->_envelope.val(), timepos_t (offset.samples()), other->len_as_tpos ())))
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, _automatable (other->session(), Temporal::AudioTime)
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, _fade_in_suspended (0)
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, _fade_out_suspended (0)
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{
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/* don't use init here, because we got fade in/out from the other region
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*/
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register_properties ();
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listen_to_my_curves ();
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connect_to_analysis_changed ();
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connect_to_header_position_offset_changed ();
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assert(_type == DataType::AUDIO);
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assert (_sources.size() == _master_sources.size());
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}
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AudioRegion::AudioRegion (std::shared_ptr<const AudioRegion> other, const SourceList& srcs)
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: Region (std::static_pointer_cast<const Region>(other), srcs)
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, AUDIOREGION_COPY_STATE (other)
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, _envelope (Properties::envelope, std::shared_ptr<AutomationList> (new AutomationList (*other->_envelope.val())))
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, _automatable (other->session(), Temporal::AudioTime)
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, _fade_in_suspended (0)
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, _fade_out_suspended (0)
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{
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/* make-a-sort-of-copy-with-different-sources constructor (used by audio filter) */
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register_properties ();
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listen_to_my_curves ();
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connect_to_analysis_changed ();
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connect_to_header_position_offset_changed ();
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assert (_sources.size() == _master_sources.size());
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}
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AudioRegion::AudioRegion (SourceList& srcs)
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: Region (srcs)
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, AUDIOREGION_STATE_DEFAULT
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, _envelope (Properties::envelope, std::shared_ptr<AutomationList> (new AutomationList(Evoral::Parameter(EnvelopeAutomation), Temporal::AudioTime)))
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, _automatable(srcs[0]->session(), Temporal::AudioTime)
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, _fade_in_suspended (0)
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, _fade_out_suspended (0)
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{
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init ();
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assert(_type == DataType::AUDIO);
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assert (_sources.size() == _master_sources.size());
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}
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AudioRegion::~AudioRegion ()
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{
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}
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void
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AudioRegion::post_set (const PropertyChange& /*ignored*/)
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{
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if (!_sync_marked) {
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_sync_position = _start;
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}
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/* return to default fades if the existing ones are too long */
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if (_left_of_split) {
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if (_fade_in->when(false) >= len_as_tpos ()) {
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set_default_fade_in ();
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}
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set_default_fade_out ();
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_left_of_split = false;
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}
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if (_right_of_split) {
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if (_fade_out->when(false) >= len_as_tpos ()) {
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set_default_fade_out ();
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}
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set_default_fade_in ();
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_right_of_split = false;
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}
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/* If _length changed, adjust our gain envelope accordingly */
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_envelope->truncate_end (len_as_tpos ());
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}
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void
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AudioRegion::connect_to_analysis_changed ()
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{
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for (SourceList::const_iterator i = _sources.begin(); i != _sources.end(); ++i) {
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(*i)->AnalysisChanged.connect_same_thread (*this, boost::bind (&AudioRegion::maybe_invalidate_transients, this));
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}
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}
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void
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AudioRegion::connect_to_header_position_offset_changed ()
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{
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set<std::shared_ptr<Source> > unique_srcs;
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for (SourceList::const_iterator i = _sources.begin(); i != _sources.end(); ++i) {
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/* connect only once to HeaderPositionOffsetChanged, even if sources are replicated
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*/
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if (unique_srcs.find (*i) == unique_srcs.end ()) {
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unique_srcs.insert (*i);
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std::shared_ptr<AudioFileSource> afs = std::dynamic_pointer_cast<AudioFileSource> (*i);
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if (afs) {
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afs->HeaderPositionOffsetChanged.connect_same_thread (*this, boost::bind (&AudioRegion::source_offset_changed, this));
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}
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}
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}
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}
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void
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AudioRegion::listen_to_my_curves ()
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{
|
|
_envelope->StateChanged.connect_same_thread (*this, boost::bind (&AudioRegion::envelope_changed, this));
|
|
_fade_in->StateChanged.connect_same_thread (*this, boost::bind (&AudioRegion::fade_in_changed, this));
|
|
_fade_out->StateChanged.connect_same_thread (*this, boost::bind (&AudioRegion::fade_out_changed, this));
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_envelope_active (bool yn)
|
|
{
|
|
if (envelope_active() != yn) {
|
|
_envelope_active = yn;
|
|
send_change (PropertyChange (Properties::envelope_active));
|
|
}
|
|
}
|
|
|
|
/** @param buf Buffer to put peak data in.
|
|
* @param npeaks Number of peaks to read (ie the number of PeakDatas in buf)
|
|
* @param offset Start position, as an offset from the start of this region's source.
|
|
* @param cnt Number of samples to read.
|
|
* @param chan_n Channel.
|
|
* @param samples_per_pixel Number of samples to use to generate one peak value.
|
|
*/
|
|
|
|
ARDOUR::samplecnt_t
|
|
AudioRegion::read_peaks (PeakData *buf, samplecnt_t npeaks, samplecnt_t offset, samplecnt_t cnt, uint32_t chan_n, double samples_per_pixel) const
|
|
{
|
|
if (chan_n >= _sources.size()) {
|
|
return 0;
|
|
}
|
|
|
|
if (audio_source(chan_n)->read_peaks (buf, npeaks, offset, cnt, samples_per_pixel)) {
|
|
return 0;
|
|
}
|
|
|
|
if (_scale_amplitude < 0.f) {
|
|
for (samplecnt_t n = 0; n < npeaks; ++n) {
|
|
const float tmp = buf[n].max;
|
|
buf[n].max = _scale_amplitude * buf[n].min;
|
|
buf[n].min = _scale_amplitude * tmp;
|
|
}
|
|
} else if (_scale_amplitude != 1.0f) {
|
|
for (samplecnt_t n = 0; n < npeaks; ++n) {
|
|
buf[n].max *= _scale_amplitude;
|
|
buf[n].min *= _scale_amplitude;
|
|
}
|
|
}
|
|
|
|
return npeaks;
|
|
}
|
|
|
|
/** @param buf Buffer to write data to (existing data will be overwritten).
|
|
* @param pos Position to read from as an offset from the region position.
|
|
* @param cnt Number of samples to read.
|
|
* @param channel Channel to read from.
|
|
*/
|
|
samplecnt_t
|
|
AudioRegion::read (Sample* buf, samplepos_t pos, samplecnt_t cnt, int channel) const
|
|
{
|
|
/* raw read, no fades, no gain, nada */
|
|
return read_from_sources (_sources, _length.val().samples(), buf, position().samples() + pos, cnt, channel);
|
|
}
|
|
|
|
samplecnt_t
|
|
AudioRegion::master_read_at (Sample *buf, Sample* /*mixdown_buffer*/, float* /*gain_buffer*/,
|
|
samplepos_t position, samplecnt_t cnt, uint32_t chan_n) const
|
|
{
|
|
/* do not read gain/scaling/fades and do not count this disk i/o in statistics */
|
|
|
|
assert (cnt >= 0);
|
|
return read_from_sources (_master_sources, _master_sources.front()->length ().samples(), buf, position, cnt, chan_n);
|
|
}
|
|
|
|
/** @param buf Buffer to mix data into.
|
|
* @param mixdown_buffer Scratch buffer for audio data.
|
|
* @param gain_buffer Scratch buffer for gain data.
|
|
* @param pos Position within the session to read from.
|
|
* @param cnt Number of samples to read.
|
|
* @param chan_n Channel number to read.
|
|
*/
|
|
samplecnt_t
|
|
AudioRegion::read_at (Sample *buf, Sample *mixdown_buffer, float *gain_buffer,
|
|
samplepos_t pos,
|
|
samplecnt_t cnt,
|
|
uint32_t chan_n) const
|
|
{
|
|
/* We are reading data from this region into buf (possibly via mixdown_buffer).
|
|
The caller has verified that we cover the desired section.
|
|
*/
|
|
|
|
/* See doc/region_read.svg for a drawing which might help to explain
|
|
what is going on.
|
|
*/
|
|
|
|
assert (cnt >= 0);
|
|
|
|
if (n_channels() == 0) {
|
|
return 0;
|
|
}
|
|
|
|
/* WORK OUT WHERE TO GET DATA FROM */
|
|
|
|
samplecnt_t to_read;
|
|
const samplepos_t psamples = position().samples();
|
|
const samplecnt_t lsamples = _length.val().samples();
|
|
|
|
assert (pos >= psamples);
|
|
sampleoffset_t const internal_offset = pos - psamples;
|
|
|
|
if (internal_offset >= lsamples) {
|
|
return 0; /* read nothing */
|
|
}
|
|
|
|
if ((to_read = min (cnt, lsamples - internal_offset)) == 0) {
|
|
return 0; /* read nothing */
|
|
}
|
|
|
|
std::shared_ptr<Playlist> pl (playlist());
|
|
if (!pl){
|
|
return 0;
|
|
}
|
|
|
|
/* COMPUTE DETAILS OF ANY FADES INVOLVED IN THIS READ */
|
|
|
|
/* Amount (length) of fade in that we are dealing with in this read */
|
|
samplecnt_t fade_in_limit = 0;
|
|
|
|
/* Offset from buf / mixdown_buffer of the start
|
|
of any fade out that we are dealing with
|
|
*/
|
|
sampleoffset_t fade_out_offset = 0;
|
|
|
|
/* Amount (length) of fade out that we are dealing with in this read */
|
|
samplecnt_t fade_out_limit = 0;
|
|
|
|
samplecnt_t fade_interval_start = 0;
|
|
|
|
/* Fade in */
|
|
|
|
if (_fade_in_active && _session.config.get_use_region_fades()) {
|
|
|
|
samplecnt_t fade_in_length = _fade_in->when(false).samples();
|
|
|
|
/* see if this read is within the fade in */
|
|
|
|
if (internal_offset < fade_in_length) {
|
|
fade_in_limit = min (to_read, fade_in_length - internal_offset);
|
|
}
|
|
}
|
|
|
|
/* Fade out */
|
|
|
|
if (_fade_out_active && _session.config.get_use_region_fades()) {
|
|
|
|
/* see if some part of this read is within the fade out */
|
|
|
|
/* ................. >| REGION
|
|
* _length
|
|
*
|
|
* { } FADE
|
|
* fade_out_length
|
|
* ^
|
|
* _length - fade_out_length
|
|
*
|
|
* |--------------|
|
|
* ^internal_offset
|
|
* ^internal_offset + to_read
|
|
*
|
|
* we need the intersection of [internal_offset,internal_offset+to_read] with
|
|
* [_length - fade_out_length, _length]
|
|
*
|
|
*/
|
|
|
|
fade_interval_start = max (internal_offset, lsamples - _fade_out->when(false).samples());
|
|
samplecnt_t fade_interval_end = min(internal_offset + to_read, lsamples);
|
|
|
|
if (fade_interval_end > fade_interval_start) {
|
|
/* (part of the) the fade out is in this buffer */
|
|
fade_out_limit = fade_interval_end - fade_interval_start;
|
|
fade_out_offset = fade_interval_start - internal_offset;
|
|
}
|
|
}
|
|
|
|
/* READ DATA FROM THE SOURCE INTO mixdown_buffer.
|
|
We can never read directly into buf, since it may contain data
|
|
from a region `below' this one in the stack, and our fades (if they exist)
|
|
may need to mix with the existing data.
|
|
*/
|
|
|
|
if (read_from_sources (_sources, lsamples, mixdown_buffer, pos, to_read, chan_n) != to_read) {
|
|
return 0;
|
|
}
|
|
|
|
/* APPLY REGULAR GAIN CURVES AND SCALING TO mixdown_buffer */
|
|
|
|
if (envelope_active()) {
|
|
_envelope->curve().get_vector (timepos_t (internal_offset), timepos_t (internal_offset + to_read), gain_buffer, to_read);
|
|
|
|
if (_scale_amplitude != 1.0f) {
|
|
for (samplecnt_t n = 0; n < to_read; ++n) {
|
|
mixdown_buffer[n] *= gain_buffer[n] * _scale_amplitude;
|
|
}
|
|
} else {
|
|
for (samplecnt_t n = 0; n < to_read; ++n) {
|
|
mixdown_buffer[n] *= gain_buffer[n];
|
|
}
|
|
}
|
|
} else if (_scale_amplitude != 1.0f) {
|
|
apply_gain_to_buffer (mixdown_buffer, to_read, _scale_amplitude);
|
|
}
|
|
|
|
/* APPLY FADES TO THE DATA IN mixdown_buffer AND MIX THE RESULTS INTO
|
|
* buf. The key things to realize here: (1) the fade being applied is
|
|
* (as of April 26th 2012) just the inverse of the fade in curve (2)
|
|
* "buf" contains data from lower regions already. So this operation
|
|
* fades out the existing material.
|
|
*/
|
|
|
|
bool is_opaque = opaque();
|
|
|
|
if (fade_in_limit != 0) {
|
|
|
|
if (is_opaque) {
|
|
if (_inverse_fade_in) {
|
|
|
|
/* explicit inverse fade in curve (e.g. for constant
|
|
* power), so we have to fetch it.
|
|
*/
|
|
|
|
_inverse_fade_in->curve().get_vector (timepos_t (internal_offset), timepos_t (internal_offset + fade_in_limit), gain_buffer, fade_in_limit);
|
|
|
|
/* Fade the data from lower layers out */
|
|
for (samplecnt_t n = 0; n < fade_in_limit; ++n) {
|
|
buf[n] *= gain_buffer[n];
|
|
}
|
|
|
|
/* refill gain buffer with the fade in */
|
|
|
|
_fade_in->curve().get_vector (timepos_t (internal_offset), timepos_t (internal_offset + fade_in_limit), gain_buffer, fade_in_limit);
|
|
|
|
} else {
|
|
|
|
/* no explicit inverse fade in, so just use (1 - fade
|
|
* in) for the fade out of lower layers
|
|
*/
|
|
|
|
_fade_in->curve().get_vector (timepos_t (internal_offset), timepos_t (internal_offset + fade_in_limit), gain_buffer, fade_in_limit);
|
|
|
|
for (samplecnt_t n = 0; n < fade_in_limit; ++n) {
|
|
buf[n] *= 1 - gain_buffer[n];
|
|
}
|
|
}
|
|
} else {
|
|
_fade_in->curve().get_vector (timepos_t (internal_offset), timepos_t (internal_offset + fade_in_limit), gain_buffer, fade_in_limit);
|
|
}
|
|
|
|
/* Mix our newly-read data in, with the fade */
|
|
for (samplecnt_t n = 0; n < fade_in_limit; ++n) {
|
|
buf[n] += mixdown_buffer[n] * gain_buffer[n];
|
|
}
|
|
}
|
|
|
|
if (fade_out_limit != 0) {
|
|
|
|
samplecnt_t const curve_offset = fade_interval_start - _fade_out->when(false).distance (len_as_tpos ()).samples();
|
|
|
|
if (is_opaque) {
|
|
if (_inverse_fade_out) {
|
|
|
|
_inverse_fade_out->curve().get_vector (timepos_t (curve_offset), timepos_t (curve_offset + fade_out_limit), gain_buffer, fade_out_limit);
|
|
|
|
/* Fade the data from lower levels in */
|
|
for (samplecnt_t n = 0, m = fade_out_offset; n < fade_out_limit; ++n, ++m) {
|
|
buf[m] *= gain_buffer[n];
|
|
}
|
|
|
|
/* fetch the actual fade out */
|
|
|
|
_fade_out->curve().get_vector (timepos_t (curve_offset), timepos_t (curve_offset + fade_out_limit), gain_buffer, fade_out_limit);
|
|
|
|
} else {
|
|
|
|
/* no explicit inverse fade out (which is
|
|
* actually a fade in), so just use (1 - fade
|
|
* out) for the fade in of lower layers
|
|
*/
|
|
|
|
_fade_out->curve().get_vector (timepos_t (curve_offset), timepos_t (curve_offset + fade_out_limit), gain_buffer, fade_out_limit);
|
|
|
|
for (samplecnt_t n = 0, m = fade_out_offset; n < fade_out_limit; ++n, ++m) {
|
|
buf[m] *= 1 - gain_buffer[n];
|
|
}
|
|
}
|
|
} else {
|
|
_fade_out->curve().get_vector (timepos_t (curve_offset), timepos_t (curve_offset + fade_out_limit), gain_buffer, fade_out_limit);
|
|
}
|
|
|
|
/* Mix our newly-read data with whatever was already there,
|
|
with the fade out applied to our data.
|
|
*/
|
|
for (samplecnt_t n = 0, m = fade_out_offset; n < fade_out_limit; ++n, ++m) {
|
|
buf[m] += mixdown_buffer[m] * gain_buffer[n];
|
|
}
|
|
}
|
|
|
|
/* MIX OR COPY THE REGION BODY FROM mixdown_buffer INTO buf */
|
|
|
|
samplecnt_t const N = to_read - fade_in_limit - fade_out_limit;
|
|
|
|
if (N > 0) {
|
|
if (is_opaque) {
|
|
DEBUG_TRACE (DEBUG::AudioPlayback, string_compose ("Region %1 memcpy into buf @ %2 + %3, from mixdown buffer @ %4 + %5, len = %6 cnt was %7\n",
|
|
name(), buf, fade_in_limit, mixdown_buffer, fade_in_limit, N, cnt));
|
|
memcpy (buf + fade_in_limit, mixdown_buffer + fade_in_limit, N * sizeof (Sample));
|
|
} else {
|
|
mix_buffers_no_gain (buf + fade_in_limit, mixdown_buffer + fade_in_limit, N);
|
|
}
|
|
}
|
|
|
|
return to_read;
|
|
}
|
|
|
|
/** Read data directly from one of our sources, accounting for the situation when the track has a different channel
|
|
* count to the region.
|
|
*
|
|
* @param srcs Source list to get our source from.
|
|
* @param limit Furthest that we should read, as an offset from the region position.
|
|
* @param buf Buffer to write data into (existing contents of the buffer will be overwritten)
|
|
* @param pos Position to read from, in session samples.
|
|
* @param cnt Number of samples to read.
|
|
* @param chan_n Channel to read from.
|
|
* @return Number of samples read.
|
|
*/
|
|
|
|
samplecnt_t
|
|
AudioRegion::read_from_sources (SourceList const & srcs, samplecnt_t limit, Sample* buf, samplepos_t pos, samplecnt_t cnt, uint32_t chan_n) const
|
|
{
|
|
sampleoffset_t const internal_offset = pos - position().samples();
|
|
|
|
if (internal_offset >= limit) {
|
|
return 0;
|
|
}
|
|
|
|
samplecnt_t const to_read = min (cnt, limit - internal_offset);
|
|
if (to_read == 0) {
|
|
return 0;
|
|
}
|
|
|
|
if (chan_n < n_channels()) {
|
|
|
|
std::shared_ptr<AudioSource> src = std::dynamic_pointer_cast<AudioSource> (srcs[chan_n]);
|
|
|
|
if (src->read (buf, _start.val().samples() + internal_offset, to_read) != to_read) {
|
|
return 0; /* "read nothing" */
|
|
}
|
|
|
|
} else {
|
|
|
|
/* track is N-channel, this region has fewer channels; silence the ones
|
|
we don't have.
|
|
*/
|
|
|
|
if (Config->get_replicate_missing_region_channels()) {
|
|
|
|
/* copy an existing channel's data in for this non-existant one */
|
|
|
|
uint32_t channel = chan_n % n_channels();
|
|
std::shared_ptr<AudioSource> src = std::dynamic_pointer_cast<AudioSource> (srcs[channel]);
|
|
|
|
if (src->read (buf, _start.val().samples() + internal_offset, to_read) != to_read) {
|
|
return 0; /* "read nothing" */
|
|
}
|
|
|
|
} else {
|
|
|
|
/* use silence */
|
|
memset (buf, 0, sizeof (Sample) * to_read);
|
|
}
|
|
}
|
|
|
|
return to_read;
|
|
}
|
|
|
|
XMLNode&
|
|
AudioRegion::get_basic_state () const
|
|
{
|
|
XMLNode& node (Region::state ());
|
|
|
|
node.set_property ("channels", (uint32_t)_sources.size());
|
|
|
|
return node;
|
|
}
|
|
|
|
XMLNode&
|
|
AudioRegion::state () const
|
|
{
|
|
XMLNode& node (get_basic_state());
|
|
XMLNode *child;
|
|
|
|
child = node.add_child ("Envelope");
|
|
|
|
bool default_env = false;
|
|
|
|
// If there are only two points, the points are in the start of the region and the end of the region
|
|
// so, if they are both at 1.0f, that means the default region.
|
|
|
|
if (_envelope->size() == 2 &&
|
|
_envelope->front()->value == GAIN_COEFF_UNITY &&
|
|
_envelope->back()->value==GAIN_COEFF_UNITY) {
|
|
if (_envelope->front()->when == 0 && _envelope->back()->when == len_as_tpos ()) {
|
|
default_env = true;
|
|
}
|
|
}
|
|
|
|
if (default_env) {
|
|
child->set_property ("default", "yes");
|
|
} else {
|
|
child->add_child_nocopy (_envelope->get_state ());
|
|
}
|
|
|
|
child = node.add_child (X_("FadeIn"));
|
|
|
|
if (_default_fade_in) {
|
|
child->set_property ("default", "yes");
|
|
} else {
|
|
child->add_child_nocopy (_fade_in->get_state ());
|
|
}
|
|
|
|
if (_inverse_fade_in) {
|
|
child = node.add_child (X_("InverseFadeIn"));
|
|
child->add_child_nocopy (_inverse_fade_in->get_state ());
|
|
}
|
|
|
|
child = node.add_child (X_("FadeOut"));
|
|
|
|
if (_default_fade_out) {
|
|
child->set_property ("default", "yes");
|
|
} else {
|
|
child->add_child_nocopy (_fade_out->get_state ());
|
|
}
|
|
|
|
if (_inverse_fade_out) {
|
|
child = node.add_child (X_("InverseFadeOut"));
|
|
child->add_child_nocopy (_inverse_fade_out->get_state ());
|
|
}
|
|
|
|
return node;
|
|
}
|
|
|
|
int
|
|
AudioRegion::_set_state (const XMLNode& node, int version, PropertyChange& what_changed, bool send)
|
|
{
|
|
const XMLNodeList& nlist = node.children();
|
|
std::shared_ptr<Playlist> the_playlist (_playlist.lock());
|
|
|
|
suspend_property_changes ();
|
|
|
|
if (the_playlist) {
|
|
the_playlist->freeze ();
|
|
}
|
|
|
|
|
|
/* this will set all our State members and stuff controlled by the Region.
|
|
It should NOT send any changed signals - that is our responsibility.
|
|
*/
|
|
|
|
Region::_set_state (node, version, what_changed, false);
|
|
|
|
float val;
|
|
if (node.get_property ("scale-gain", val)) {
|
|
if (val != _scale_amplitude) {
|
|
_scale_amplitude = val;
|
|
what_changed.add (Properties::scale_amplitude);
|
|
}
|
|
}
|
|
|
|
/* Now find envelope description and other related child items */
|
|
|
|
_envelope->freeze ();
|
|
|
|
for (XMLNodeConstIterator niter = nlist.begin(); niter != nlist.end(); ++niter) {
|
|
XMLNode *child;
|
|
XMLProperty const * prop;
|
|
|
|
child = (*niter);
|
|
|
|
if (child->name() == "Envelope") {
|
|
|
|
_envelope->clear ();
|
|
|
|
if ((prop = child->property ("default")) != 0 || _envelope->set_state (*child, version)) {
|
|
set_default_envelope ();
|
|
}
|
|
|
|
_envelope->truncate_end (len_as_tpos ());
|
|
|
|
|
|
} else if (child->name() == "FadeIn") {
|
|
|
|
_fade_in->clear ();
|
|
|
|
bool is_default;
|
|
if ((child->get_property ("default", is_default) && is_default) || (prop = child->property ("steepness")) != 0) {
|
|
set_default_fade_in ();
|
|
} else {
|
|
XMLNode* grandchild = child->child ("AutomationList");
|
|
if (grandchild) {
|
|
_fade_in->set_state (*grandchild, version);
|
|
}
|
|
}
|
|
|
|
bool is_active;
|
|
if (child->get_property ("active", is_active)) {
|
|
set_fade_in_active (is_active);
|
|
}
|
|
|
|
} else if (child->name() == "FadeOut") {
|
|
|
|
_fade_out->clear ();
|
|
|
|
bool is_default;
|
|
if ((child->get_property ("default", is_default) && is_default) || (prop = child->property ("steepness")) != 0) {
|
|
set_default_fade_out ();
|
|
} else {
|
|
XMLNode* grandchild = child->child ("AutomationList");
|
|
if (grandchild) {
|
|
_fade_out->set_state (*grandchild, version);
|
|
}
|
|
}
|
|
|
|
bool is_active;
|
|
if (child->get_property ("active", is_active)) {
|
|
set_fade_out_active (is_active);
|
|
}
|
|
|
|
} else if ( (child->name() == "InverseFadeIn") || (child->name() == "InvFadeIn") ) {
|
|
XMLNode* grandchild = child->child ("AutomationList");
|
|
if (grandchild) {
|
|
_inverse_fade_in->set_state (*grandchild, version);
|
|
}
|
|
} else if ( (child->name() == "InverseFadeOut") || (child->name() == "InvFadeOut") ) {
|
|
XMLNode* grandchild = child->child ("AutomationList");
|
|
if (grandchild) {
|
|
_inverse_fade_out->set_state (*grandchild, version);
|
|
}
|
|
}
|
|
}
|
|
|
|
_envelope->thaw ();
|
|
resume_property_changes ();
|
|
|
|
if (send) {
|
|
send_change (what_changed);
|
|
}
|
|
|
|
if (the_playlist) {
|
|
the_playlist->thaw ();
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
AudioRegion::set_state (const XMLNode& node, int version)
|
|
{
|
|
PropertyChange what_changed;
|
|
return _set_state (node, version, what_changed, true);
|
|
}
|
|
|
|
void
|
|
AudioRegion::fade_range (samplepos_t start, samplepos_t end)
|
|
{
|
|
samplepos_t s, e;
|
|
|
|
switch (coverage (timepos_t (start), timepos_t (end))) {
|
|
case Temporal::OverlapStart:
|
|
trim_front (timepos_t (start));
|
|
s = position().samples();
|
|
e = end;
|
|
set_fade_in (FadeConstantPower, e - s);
|
|
break;
|
|
case Temporal::OverlapEnd:
|
|
trim_end(timepos_t (end));
|
|
s = start;
|
|
e = (position() + timepos_t (_length)).samples();
|
|
set_fade_out (FadeConstantPower, e - s);
|
|
break;
|
|
case Temporal::OverlapInternal:
|
|
/* needs addressing, perhaps. Difficult to do if we can't
|
|
* control one edge of the fade relative to the relevant edge
|
|
* of the region, which we cannot - fades are currently assumed
|
|
* to start/end at the start/end of the region
|
|
*/
|
|
break;
|
|
default:
|
|
return;
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_fade_in_shape (FadeShape shape)
|
|
{
|
|
set_fade_in (shape, _fade_in->when(false).samples());
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_fade_out_shape (FadeShape shape)
|
|
{
|
|
set_fade_out (shape, _fade_out->when(false).samples());
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_fade_in (std::shared_ptr<AutomationList> f)
|
|
{
|
|
_fade_in->freeze ();
|
|
*(_fade_in.val()) = *f;
|
|
_fade_in->thaw ();
|
|
_default_fade_in = false;
|
|
|
|
send_change (PropertyChange (Properties::fade_in));
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_fade_in (FadeShape shape, samplecnt_t len)
|
|
{
|
|
const ARDOUR::ParameterDescriptor desc(FadeInAutomation);
|
|
std::shared_ptr<Evoral::ControlList> c1 (new Evoral::ControlList (FadeInAutomation, desc, Temporal::AudioTime));
|
|
std::shared_ptr<Evoral::ControlList> c2 (new Evoral::ControlList (FadeInAutomation, desc, Temporal::AudioTime));
|
|
std::shared_ptr<Evoral::ControlList> c3 (new Evoral::ControlList (FadeInAutomation, desc, Temporal::AudioTime));
|
|
|
|
_fade_in->freeze ();
|
|
_fade_in->clear ();
|
|
_inverse_fade_in->clear ();
|
|
|
|
const int num_steps = 32;
|
|
|
|
switch (shape) {
|
|
case FadeLinear:
|
|
_fade_in->fast_simple_add (timepos_t (Temporal::AudioTime), GAIN_COEFF_SMALL);
|
|
_fade_in->fast_simple_add (timepos_t ((samplepos_t)len), GAIN_COEFF_UNITY);
|
|
reverse_curve (_inverse_fade_in.val(), _fade_in.val());
|
|
break;
|
|
|
|
case FadeFast:
|
|
generate_db_fade (_fade_in.val(), len, num_steps, -60);
|
|
reverse_curve (c1, _fade_in.val());
|
|
_fade_in->copy_events (*c1);
|
|
generate_inverse_power_curve (_inverse_fade_in.val(), _fade_in.val());
|
|
break;
|
|
|
|
case FadeSlow:
|
|
generate_db_fade (c1, len, num_steps, -1); // start off with a slow fade
|
|
generate_db_fade (c2, len, num_steps, -80); // end with a fast fade
|
|
merge_curves (_fade_in.val(), c1, c2);
|
|
reverse_curve (c3, _fade_in.val());
|
|
_fade_in->copy_events (*c3);
|
|
generate_inverse_power_curve (_inverse_fade_in.val(), _fade_in.val());
|
|
break;
|
|
|
|
case FadeConstantPower:
|
|
_fade_in->fast_simple_add (timepos_t (Temporal::AudioTime), GAIN_COEFF_SMALL);
|
|
for (int i = 1; i < num_steps; ++i) {
|
|
const float dist = i / (num_steps + 1.f);
|
|
_fade_in->fast_simple_add (timepos_t ((samplepos_t)(len * dist)), sin (dist * M_PI / 2.0));
|
|
}
|
|
_fade_in->fast_simple_add (timepos_t ((samplepos_t)len), GAIN_COEFF_UNITY);
|
|
reverse_curve (_inverse_fade_in.val(), _fade_in.val());
|
|
break;
|
|
|
|
case FadeSymmetric:
|
|
//start with a nearly linear cuve
|
|
_fade_in->fast_simple_add (timepos_t (Temporal::AudioTime), 1);
|
|
_fade_in->fast_simple_add (timepos_t ((samplepos_t)(0.5 * len)), 0.6);
|
|
//now generate a fade-out curve by successively applying a gain drop
|
|
const double breakpoint = 0.7; //linear for first 70%
|
|
for (int i = 2; i < 9; ++i) {
|
|
const float coeff = (1.f - breakpoint) * powf (0.5, i);
|
|
_fade_in->fast_simple_add (timepos_t ((samplepos_t)(len * (breakpoint + ((GAIN_COEFF_UNITY - breakpoint) * (double)i / 9.0)))), coeff);
|
|
}
|
|
_fade_in->fast_simple_add (timepos_t ((samplepos_t)len), GAIN_COEFF_SMALL);
|
|
reverse_curve (c3, _fade_in.val());
|
|
_fade_in->copy_events (*c3);
|
|
reverse_curve (_inverse_fade_in.val(), _fade_in.val());
|
|
break;
|
|
}
|
|
|
|
_fade_in->set_interpolation(Evoral::ControlList::Curved);
|
|
_inverse_fade_in->set_interpolation(Evoral::ControlList::Curved);
|
|
|
|
_default_fade_in = false;
|
|
_fade_in->thaw ();
|
|
send_change (PropertyChange (Properties::fade_in));
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_fade_out (std::shared_ptr<AutomationList> f)
|
|
{
|
|
_fade_out->freeze ();
|
|
*(_fade_out.val()) = *f;
|
|
_fade_out->thaw ();
|
|
_default_fade_out = false;
|
|
|
|
send_change (PropertyChange (Properties::fade_out));
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_fade_out (FadeShape shape, samplecnt_t len)
|
|
{
|
|
const ARDOUR::ParameterDescriptor desc(FadeOutAutomation);
|
|
std::shared_ptr<Evoral::ControlList> c1 (new Evoral::ControlList (FadeOutAutomation, desc, Temporal::AudioTime));
|
|
std::shared_ptr<Evoral::ControlList> c2 (new Evoral::ControlList (FadeOutAutomation, desc, Temporal::AudioTime));
|
|
|
|
_fade_out->freeze ();
|
|
_fade_out->clear ();
|
|
_inverse_fade_out->clear ();
|
|
|
|
const int num_steps = 32;
|
|
|
|
switch (shape) {
|
|
case FadeLinear:
|
|
_fade_out->fast_simple_add (timepos_t (Temporal::AudioTime), GAIN_COEFF_UNITY);
|
|
_fade_out->fast_simple_add (timepos_t ((samplepos_t)len), GAIN_COEFF_SMALL);
|
|
reverse_curve (_inverse_fade_out.val(), _fade_out.val());
|
|
break;
|
|
|
|
case FadeFast:
|
|
generate_db_fade (_fade_out.val(), len, num_steps, -60);
|
|
generate_inverse_power_curve (_inverse_fade_out.val(), _fade_out.val());
|
|
break;
|
|
|
|
case FadeSlow:
|
|
generate_db_fade (c1, len, num_steps, -1); //start off with a slow fade
|
|
generate_db_fade (c2, len, num_steps, -80); //end with a fast fade
|
|
merge_curves (_fade_out.val(), c1, c2);
|
|
generate_inverse_power_curve (_inverse_fade_out.val(), _fade_out.val());
|
|
break;
|
|
|
|
case FadeConstantPower:
|
|
//constant-power fades use a sin/cos relationship
|
|
//the cutoff is abrupt but it has the benefit of being symmetrical
|
|
_fade_out->fast_simple_add (timepos_t (Temporal::AudioTime), GAIN_COEFF_UNITY);
|
|
for (int i = 1; i < num_steps; ++i) {
|
|
const float dist = i / (num_steps + 1.f);
|
|
_fade_out->fast_simple_add (timepos_t ((samplepos_t)(len * dist)), cos (dist * M_PI / 2.0));
|
|
}
|
|
_fade_out->fast_simple_add (timepos_t (len), GAIN_COEFF_SMALL);
|
|
reverse_curve (_inverse_fade_out.val(), _fade_out.val());
|
|
break;
|
|
|
|
case FadeSymmetric:
|
|
//start with a nearly linear cuve
|
|
_fade_out->fast_simple_add (timepos_t (Temporal::AudioTime), 1);
|
|
_fade_out->fast_simple_add (timepos_t ((samplepos_t)(0.5 * len)), 0.6);
|
|
//now generate a fade-out curve by successively applying a gain drop
|
|
const double breakpoint = 0.7; //linear for first 70%
|
|
for (int i = 2; i < 9; ++i) {
|
|
const float coeff = (1.f - breakpoint) * powf (0.5, i);
|
|
_fade_out->fast_simple_add (timepos_t ((samplepos_t)(len * (breakpoint + ((GAIN_COEFF_UNITY - breakpoint) * (double)i / 9.0)))), coeff);
|
|
}
|
|
_fade_out->fast_simple_add (timepos_t ((samplepos_t)len), GAIN_COEFF_SMALL);
|
|
reverse_curve (_inverse_fade_out.val(), _fade_out.val());
|
|
break;
|
|
}
|
|
|
|
_fade_out->set_interpolation(Evoral::ControlList::Curved);
|
|
_inverse_fade_out->set_interpolation(Evoral::ControlList::Curved);
|
|
|
|
_default_fade_out = false;
|
|
_fade_out->thaw ();
|
|
send_change (PropertyChange (Properties::fade_out));
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_fade_in_length (samplecnt_t len)
|
|
{
|
|
if (len > length_samples()) {
|
|
len = length_samples() - 1;
|
|
}
|
|
|
|
if (len < 64) {
|
|
len = 64;
|
|
}
|
|
|
|
timepos_t const tlen = timepos_t ((samplepos_t)len);
|
|
|
|
bool changed = _fade_in->extend_to (tlen);
|
|
|
|
if (changed) {
|
|
if (_inverse_fade_in) {
|
|
_inverse_fade_in->extend_to (tlen);
|
|
}
|
|
|
|
_default_fade_in = false;
|
|
send_change (PropertyChange (Properties::fade_in));
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_fade_out_length (samplecnt_t len)
|
|
{
|
|
if (len > length_samples()) {
|
|
len = length_samples() - 1;
|
|
}
|
|
|
|
if (len < 64) {
|
|
len = 64;
|
|
}
|
|
|
|
timepos_t const tlen = timepos_t ((samplepos_t)len);
|
|
|
|
bool changed = _fade_out->extend_to (tlen);
|
|
|
|
if (changed) {
|
|
|
|
if (_inverse_fade_out) {
|
|
_inverse_fade_out->extend_to (tlen);
|
|
}
|
|
_default_fade_out = false;
|
|
|
|
send_change (PropertyChange (Properties::fade_out));
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_fade_in_active (bool yn)
|
|
{
|
|
if (yn == _fade_in_active) {
|
|
return;
|
|
}
|
|
|
|
_fade_in_active = yn;
|
|
send_change (PropertyChange (Properties::fade_in_active));
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_fade_out_active (bool yn)
|
|
{
|
|
if (yn == _fade_out_active) {
|
|
return;
|
|
}
|
|
_fade_out_active = yn;
|
|
send_change (PropertyChange (Properties::fade_out_active));
|
|
}
|
|
|
|
bool
|
|
AudioRegion::fade_in_is_default () const
|
|
{
|
|
return _fade_in->size() == 2 && _fade_in->when(true) == 0 && _fade_in->when(false).samples () == 64;
|
|
}
|
|
|
|
bool
|
|
AudioRegion::fade_out_is_default () const
|
|
{
|
|
return _fade_out->size() == 2 && _fade_out->when(true) == 0 && _fade_out->when(false).samples () == 64;
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_default_fade_in ()
|
|
{
|
|
_fade_in_suspended = 0;
|
|
set_fade_in (Config->get_default_fade_shape(), 64);
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_default_fade_out ()
|
|
{
|
|
_fade_out_suspended = 0;
|
|
set_fade_out (Config->get_default_fade_shape(), 64);
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_default_fades ()
|
|
{
|
|
set_default_fade_in ();
|
|
set_default_fade_out ();
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_default_envelope ()
|
|
{
|
|
_envelope->freeze ();
|
|
_envelope->clear ();
|
|
_envelope->fast_simple_add (timepos_t (Temporal::AudioTime), GAIN_COEFF_UNITY);
|
|
|
|
/* Force length into audio time domain. If we don't do this, the
|
|
* envelope (which uses the AudioTime domain) will have problems when
|
|
* we call its fast_simple_add() mechanism and it discovers that the
|
|
* time is not AudioTime.
|
|
*
|
|
* XXX this needs some thought
|
|
*/
|
|
|
|
_envelope->fast_simple_add (len_as_tpos (), GAIN_COEFF_UNITY);
|
|
_envelope->thaw ();
|
|
}
|
|
|
|
void
|
|
AudioRegion::recompute_at_end ()
|
|
{
|
|
/* our length has changed. recompute a new final point by interpolating
|
|
based on the the existing curve.
|
|
*/
|
|
|
|
_envelope->freeze ();
|
|
_envelope->truncate_end (len_as_tpos ());
|
|
_envelope->thaw ();
|
|
|
|
suspend_property_changes();
|
|
|
|
if (_left_of_split) {
|
|
set_default_fade_out ();
|
|
_left_of_split = false;
|
|
} else if (_fade_out->when(false) > _length) {
|
|
_fade_out->extend_to (len_as_tpos ());
|
|
send_change (PropertyChange (Properties::fade_out));
|
|
}
|
|
|
|
if (_fade_in->when(false) > _length) {
|
|
_fade_in->extend_to (len_as_tpos ());
|
|
send_change (PropertyChange (Properties::fade_in));
|
|
}
|
|
|
|
resume_property_changes();
|
|
}
|
|
|
|
void
|
|
AudioRegion::recompute_at_start ()
|
|
{
|
|
/* as above, but the shift was from the front */
|
|
|
|
_envelope->truncate_start (timecnt_t::from_samples (length().samples ()));
|
|
|
|
suspend_property_changes();
|
|
|
|
if (_right_of_split) {
|
|
set_default_fade_in ();
|
|
_right_of_split = false;
|
|
} else if (_fade_in->when(false) > len_as_tpos ()) {
|
|
_fade_in->extend_to (len_as_tpos ());
|
|
send_change (PropertyChange (Properties::fade_in));
|
|
}
|
|
|
|
if (_fade_out->when(false) > len_as_tpos ()) {
|
|
_fade_out->extend_to (len_as_tpos ());
|
|
send_change (PropertyChange (Properties::fade_out));
|
|
}
|
|
|
|
resume_property_changes();
|
|
}
|
|
|
|
int
|
|
AudioRegion::separate_by_channel (vector<std::shared_ptr<Region> >& v) const
|
|
{
|
|
SourceList srcs;
|
|
string new_name;
|
|
int n = 0;
|
|
|
|
if (_sources.size() < 2) {
|
|
return 0;
|
|
}
|
|
|
|
for (SourceList::const_iterator i = _sources.begin(); i != _sources.end(); ++i) {
|
|
srcs.clear ();
|
|
srcs.push_back (*i);
|
|
|
|
new_name = _name;
|
|
|
|
if (_sources.size() == 2) {
|
|
if (n == 0) {
|
|
new_name += "-L";
|
|
} else {
|
|
new_name += "-R";
|
|
}
|
|
} else {
|
|
new_name += '-';
|
|
new_name += ('0' + n + 1);
|
|
}
|
|
|
|
/* create a copy with just one source. prevent if from being thought of as
|
|
"whole file" even if it covers the entire source file(s).
|
|
*/
|
|
|
|
PropertyList plist (properties ());
|
|
|
|
plist.add (Properties::name, new_name);
|
|
plist.add (Properties::whole_file, true);
|
|
|
|
v.push_back(RegionFactory::create (srcs, plist));
|
|
|
|
++n;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
samplecnt_t
|
|
AudioRegion::read_raw_internal (Sample* buf, samplepos_t pos, samplecnt_t cnt, int channel) const
|
|
{
|
|
return audio_source(channel)->read (buf, pos, cnt);
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_scale_amplitude (gain_t g)
|
|
{
|
|
std::shared_ptr<Playlist> pl (playlist());
|
|
|
|
_scale_amplitude = g;
|
|
|
|
send_change (PropertyChange (Properties::scale_amplitude));
|
|
}
|
|
|
|
double
|
|
AudioRegion::maximum_amplitude (Progress* p) const
|
|
{
|
|
samplepos_t fpos = start_sample();;
|
|
samplepos_t const fend = start_sample() + length_samples();
|
|
double maxamp = 0;
|
|
|
|
samplecnt_t const blocksize = 64 * 1024;
|
|
Sample buf[blocksize];
|
|
|
|
while (fpos < fend) {
|
|
|
|
uint32_t n;
|
|
|
|
samplecnt_t const to_read = min (fend - fpos, blocksize);
|
|
|
|
for (n = 0; n < n_channels(); ++n) {
|
|
|
|
/* read it in */
|
|
|
|
if (read_raw_internal (buf, fpos, to_read, n) != to_read) {
|
|
#ifndef NDEBUG
|
|
cerr << "AudioRegion::maximum_amplitude read failed for '" << _name << "'\n";
|
|
#endif
|
|
return 0;
|
|
}
|
|
|
|
maxamp = compute_peak (buf, to_read, maxamp);
|
|
}
|
|
|
|
fpos += to_read;
|
|
if (p) {
|
|
p->set_progress (float (fpos - start_sample()) / length_samples());
|
|
if (p->cancelled ()) {
|
|
return -1;
|
|
}
|
|
}
|
|
}
|
|
|
|
return maxamp;
|
|
}
|
|
|
|
double
|
|
AudioRegion::rms (Progress* p) const
|
|
{
|
|
samplepos_t fpos = start_sample();
|
|
samplepos_t const fend = start_sample() + length_samples();
|
|
uint32_t const n_chan = n_channels ();
|
|
double rms = 0;
|
|
|
|
samplecnt_t const blocksize = 64 * 1024;
|
|
Sample buf[blocksize];
|
|
|
|
samplecnt_t total = 0;
|
|
|
|
if (n_chan == 0 || fend == fpos) {
|
|
return 0;
|
|
}
|
|
|
|
while (fpos < fend) {
|
|
samplecnt_t const to_read = min (fend - fpos, blocksize);
|
|
for (uint32_t c = 0; c < n_chan; ++c) {
|
|
if (read_raw_internal (buf, fpos, to_read, c) != to_read) {
|
|
return 0;
|
|
}
|
|
for (samplepos_t i = 0; i < to_read; ++i) {
|
|
rms += buf[i] * buf[i];
|
|
}
|
|
}
|
|
total += to_read;
|
|
fpos += to_read;
|
|
if (p) {
|
|
p->set_progress (float (fpos - start_sample()) / length_samples());
|
|
if (p->cancelled ()) {
|
|
return -1;
|
|
}
|
|
}
|
|
}
|
|
return sqrt (2. * rms / (double)(total * n_chan));
|
|
}
|
|
|
|
bool
|
|
AudioRegion::loudness (float& tp, float& i, float& s, float& m, Progress* p) const
|
|
{
|
|
ARDOUR::AnalysisGraph ag (&_session);
|
|
tp = i = s = m = -200;
|
|
|
|
ag.set_total_samples (length_samples());
|
|
ag.analyze_region (this, true, p);
|
|
|
|
if (p && p->cancelled ()) {
|
|
return false;
|
|
}
|
|
|
|
AnalysisResults const& ar (ag.results ());
|
|
if (ar.size() != 1) {
|
|
return false;
|
|
}
|
|
ExportAnalysisPtr eap (ar.begin ()->second);
|
|
|
|
if (eap->have_dbtp) {
|
|
tp = eap->truepeak;
|
|
}
|
|
if (eap->have_loudness) {
|
|
i = eap->integrated_loudness;
|
|
s = eap->max_loudness_short;
|
|
m = eap->max_loudness_momentary;
|
|
}
|
|
|
|
return eap->have_dbtp || eap->have_loudness;
|
|
}
|
|
|
|
/** Normalize using a given maximum amplitude and target, so that region
|
|
* _scale_amplitude becomes target / max_amplitude.
|
|
*/
|
|
void
|
|
AudioRegion::normalize (float max_amplitude, float target_dB)
|
|
{
|
|
gain_t target = dB_to_coefficient (target_dB);
|
|
|
|
if (target == GAIN_COEFF_UNITY) {
|
|
/* do not normalize to precisely 1.0 (0 dBFS), to avoid making it appear
|
|
that we may have clipped.
|
|
*/
|
|
target -= FLT_EPSILON;
|
|
}
|
|
|
|
if (max_amplitude < GAIN_COEFF_SMALL) {
|
|
/* don't even try */
|
|
return;
|
|
}
|
|
|
|
if (max_amplitude == target) {
|
|
/* we can't do anything useful */
|
|
return;
|
|
}
|
|
|
|
set_scale_amplitude (target / max_amplitude);
|
|
}
|
|
|
|
void
|
|
AudioRegion::fade_in_changed ()
|
|
{
|
|
send_change (PropertyChange (Properties::fade_in));
|
|
}
|
|
|
|
void
|
|
AudioRegion::fade_out_changed ()
|
|
{
|
|
send_change (PropertyChange (Properties::fade_out));
|
|
}
|
|
|
|
void
|
|
AudioRegion::envelope_changed ()
|
|
{
|
|
send_change (PropertyChange (Properties::envelope));
|
|
}
|
|
|
|
void
|
|
AudioRegion::suspend_fade_in ()
|
|
{
|
|
if (++_fade_in_suspended == 1) {
|
|
if (fade_in_is_default()) {
|
|
set_fade_in_active (false);
|
|
}
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioRegion::resume_fade_in ()
|
|
{
|
|
if (--_fade_in_suspended == 0 && _fade_in_suspended) {
|
|
set_fade_in_active (true);
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioRegion::suspend_fade_out ()
|
|
{
|
|
if (++_fade_out_suspended == 1) {
|
|
if (fade_out_is_default()) {
|
|
set_fade_out_active (false);
|
|
}
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioRegion::resume_fade_out ()
|
|
{
|
|
if (--_fade_out_suspended == 0 &&_fade_out_suspended) {
|
|
set_fade_out_active (true);
|
|
}
|
|
}
|
|
|
|
bool
|
|
AudioRegion::speed_mismatch (float sr) const
|
|
{
|
|
if (_sources.empty()) {
|
|
/* impossible, but ... */
|
|
return false;
|
|
}
|
|
|
|
float fsr = audio_source()->sample_rate();
|
|
|
|
return fsr != sr;
|
|
}
|
|
|
|
void
|
|
AudioRegion::source_offset_changed ()
|
|
{
|
|
/* XXX this fixes a crash that should not occur. It does occur
|
|
because regions are not being deleted when a session
|
|
is unloaded. That bug must be fixed.
|
|
*/
|
|
|
|
if (_sources.empty()) {
|
|
return;
|
|
}
|
|
|
|
std::shared_ptr<AudioFileSource> afs = std::dynamic_pointer_cast<AudioFileSource>(_sources.front());
|
|
}
|
|
|
|
std::shared_ptr<AudioSource>
|
|
AudioRegion::audio_source (uint32_t n) const
|
|
{
|
|
// Guaranteed to succeed (use a static cast for speed?)
|
|
return std::dynamic_pointer_cast<AudioSource>(source(n));
|
|
}
|
|
|
|
void
|
|
AudioRegion::clear_transients () // yet unused
|
|
{
|
|
_user_transients.clear ();
|
|
_valid_transients = false;
|
|
send_change (PropertyChange (Properties::valid_transients));
|
|
}
|
|
|
|
void
|
|
AudioRegion::add_transient (samplepos_t where)
|
|
{
|
|
if (where < first_sample () || where >= last_sample ()) {
|
|
return;
|
|
}
|
|
where -= position_sample();
|
|
|
|
if (!_valid_transients) {
|
|
_transient_user_start = start_sample();
|
|
_valid_transients = true;
|
|
}
|
|
sampleoffset_t offset = _transient_user_start - start_sample();;
|
|
|
|
if (where < offset) {
|
|
if (offset <= 0) {
|
|
return;
|
|
}
|
|
// region start changed (extend to front), shift points and offset
|
|
for (AnalysisFeatureList::iterator x = _transients.begin(); x != _transients.end(); ++x) {
|
|
(*x) += offset;
|
|
}
|
|
_transient_user_start -= offset;
|
|
offset = 0;
|
|
}
|
|
|
|
const samplepos_t p = where - offset;
|
|
_user_transients.push_back(p);
|
|
send_change (PropertyChange (Properties::valid_transients));
|
|
}
|
|
|
|
void
|
|
AudioRegion::update_transient (samplepos_t old_position, samplepos_t new_position)
|
|
{
|
|
bool changed = false;
|
|
if (!_onsets.empty ()) {
|
|
const samplepos_t p = old_position - position_sample();
|
|
AnalysisFeatureList::iterator x = std::find (_onsets.begin (), _onsets.end (), p);
|
|
if (x != _transients.end ()) {
|
|
(*x) = new_position - position_sample();
|
|
changed = true;
|
|
}
|
|
}
|
|
|
|
if (_valid_transients) {
|
|
const sampleoffset_t offset = position_sample() + _transient_user_start - start_sample();
|
|
const samplepos_t p = old_position - offset;
|
|
AnalysisFeatureList::iterator x = std::find (_user_transients.begin (), _user_transients.end (), p);
|
|
if (x != _transients.end ()) {
|
|
(*x) = new_position - offset;
|
|
changed = true;
|
|
}
|
|
}
|
|
|
|
if (changed) {
|
|
send_change (PropertyChange (Properties::valid_transients));
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioRegion::remove_transient (samplepos_t where)
|
|
{
|
|
bool changed = false;
|
|
if (!_onsets.empty ()) {
|
|
const samplepos_t p = where - position_sample();
|
|
AnalysisFeatureList::iterator i = std::find (_onsets.begin (), _onsets.end (), p);
|
|
if (i != _onsets.end ()) {
|
|
_onsets.erase (i);
|
|
changed = true;
|
|
}
|
|
}
|
|
|
|
if (_valid_transients) {
|
|
const samplepos_t p = where - (position_sample() + _transient_user_start - start_sample());
|
|
AnalysisFeatureList::iterator i = std::find (_user_transients.begin (), _user_transients.end (), p);
|
|
if (i != _user_transients.end ()) {
|
|
_user_transients.erase (i);
|
|
changed = true;
|
|
}
|
|
}
|
|
|
|
if (changed) {
|
|
send_change (PropertyChange (Properties::valid_transients));
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioRegion::set_onsets (AnalysisFeatureList& results)
|
|
{
|
|
_onsets.clear();
|
|
_onsets = results;
|
|
send_change (PropertyChange (Properties::valid_transients));
|
|
}
|
|
|
|
void
|
|
AudioRegion::build_transients ()
|
|
{
|
|
_transients.clear ();
|
|
_transient_analysis_start = _transient_analysis_end = 0;
|
|
|
|
std::shared_ptr<Playlist> pl = playlist();
|
|
|
|
if (!pl) {
|
|
return;
|
|
}
|
|
|
|
/* check analyzed sources first */
|
|
SourceList::iterator s;
|
|
for (s = _sources.begin() ; s != _sources.end(); ++s) {
|
|
if (!(*s)->has_been_analysed()) {
|
|
#ifndef NDEBUG
|
|
cerr << "For " << name() << " source " << (*s)->name() << " has not been analyzed\n";
|
|
#endif
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (s == _sources.end()) {
|
|
/* all sources are analyzed, merge data from each one */
|
|
for (s = _sources.begin() ; s != _sources.end(); ++s) {
|
|
|
|
/* find the set of transients within the bounds of this region */
|
|
AnalysisFeatureList::iterator low = lower_bound ((*s)->transients.begin(),
|
|
(*s)->transients.end(),
|
|
start_sample());
|
|
|
|
AnalysisFeatureList::iterator high = upper_bound ((*s)->transients.begin(),
|
|
(*s)->transients.end(),
|
|
start_sample() + length_samples());
|
|
|
|
/* and add them */
|
|
_transients.insert (_transients.end(), low, high);
|
|
}
|
|
|
|
TransientDetector::cleanup_transients (_transients, pl->session().sample_rate(), 3.0);
|
|
|
|
/* translate all transients to current position */
|
|
for (AnalysisFeatureList::iterator x = _transients.begin(); x != _transients.end(); ++x) {
|
|
(*x) -= start_sample();
|
|
}
|
|
|
|
_transient_analysis_start = start_sample();
|
|
_transient_analysis_end = start_sample() + length_samples();
|
|
return;
|
|
}
|
|
|
|
/* no existing/complete transient info */
|
|
|
|
static bool analyse_dialog_shown = false; /* global per instance of Ardour */
|
|
|
|
if (!Config->get_auto_analyse_audio()) {
|
|
if (!analyse_dialog_shown) {
|
|
pl->session().Dialog (string_compose (_("\
|
|
You have requested an operation that requires audio analysis.\n\n\
|
|
You currently have \"auto-analyse-audio\" disabled, which means \
|
|
that transient data must be generated every time it is required.\n\n\
|
|
If you are doing work that will require transient data on a \
|
|
regular basis, you should probably enable \"auto-analyse-audio\" \
|
|
in Preferences > Audio > Regions, then quit %1 and restart.\n\n\
|
|
This dialog will not display again. But you may notice a slight delay \
|
|
in this and future transient-detection operations.\n\
|
|
"), PROGRAM_NAME));
|
|
analyse_dialog_shown = true;
|
|
}
|
|
}
|
|
|
|
try {
|
|
TransientDetector t (pl->session().sample_rate());
|
|
for (uint32_t i = 0; i < n_channels(); ++i) {
|
|
|
|
AnalysisFeatureList these_results;
|
|
|
|
t.reset ();
|
|
|
|
/* this produces analysis result relative to current position
|
|
* ::read() sample 0 is at _position */
|
|
if (t.run ("", this, i, these_results)) {
|
|
return;
|
|
}
|
|
|
|
/* merge */
|
|
_transients.insert (_transients.end(), these_results.begin(), these_results.end());
|
|
}
|
|
} catch (...) {
|
|
error << string_compose(_("Transient Analysis failed for %1."), _("Audio Region")) << endmsg;
|
|
return;
|
|
}
|
|
|
|
TransientDetector::cleanup_transients (_transients, pl->session().sample_rate(), 3.0);
|
|
_transient_analysis_start = start_sample();
|
|
_transient_analysis_end = start_sample() + length_samples();
|
|
}
|
|
|
|
/* Transient analysis uses ::read() which is relative to _start,
|
|
* at the time of analysis and spans _length samples.
|
|
*
|
|
* This is true for RhythmFerret::run_analysis and the
|
|
* TransientDetector here.
|
|
*
|
|
* We store _start and length in _transient_analysis_start,
|
|
* _transient_analysis_end in case the region is trimmed or split after analysis.
|
|
*
|
|
* Various methods (most notably Playlist::find_next_transient and
|
|
* RhythmFerret::do_split_action) span multiple regions and *merge/combine*
|
|
* Analysis results.
|
|
* We therefore need to translate the analysis timestamps to absolute session-time
|
|
* and include the _position of the region.
|
|
*
|
|
* Note: we should special case the AudioRegionView. The region-view itself
|
|
* is located at _position (currently ARV subtracts _position again)
|
|
*/
|
|
void
|
|
AudioRegion::get_transients (AnalysisFeatureList& results)
|
|
{
|
|
std::shared_ptr<Playlist> pl = playlist();
|
|
if (!playlist ()) {
|
|
return;
|
|
}
|
|
|
|
Region::merge_features (results, _user_transients, position_sample() + _transient_user_start - start_sample());
|
|
|
|
if (!_onsets.empty ()) {
|
|
// onsets are invalidated when start or length changes
|
|
merge_features (results, _onsets, position_sample());
|
|
return;
|
|
}
|
|
|
|
if ((_transient_analysis_start == _transient_analysis_end)
|
|
|| _transient_analysis_start > start_sample()
|
|
|| _transient_analysis_end < start_sample() + length_samples()) {
|
|
build_transients ();
|
|
}
|
|
|
|
merge_features (results, _transients, position_sample() + _transient_analysis_start - start_sample());
|
|
}
|
|
|
|
/** Find areas of `silence' within a region.
|
|
*
|
|
* @param threshold Threshold below which signal is considered silence (as a sample value)
|
|
* @param min_length Minimum length of silent period to be reported.
|
|
* @return Silent intervals, measured relative to the region start in the source
|
|
*/
|
|
|
|
AudioIntervalResult
|
|
AudioRegion::find_silence (Sample threshold, samplecnt_t min_length, samplecnt_t fade_length, InterThreadInfo& itt) const
|
|
{
|
|
samplecnt_t const block_size = 64 * 1024;
|
|
boost::scoped_array<Sample> loudest (new Sample[block_size]);
|
|
boost::scoped_array<Sample> buf (new Sample[block_size]);
|
|
|
|
assert (fade_length >= 0);
|
|
assert (min_length > 0);
|
|
|
|
samplepos_t pos = start_sample();
|
|
samplepos_t const end = start_sample() + length_samples();
|
|
|
|
AudioIntervalResult silent_periods;
|
|
|
|
bool in_silence = true;
|
|
sampleoffset_t silence_start = start_sample();
|
|
|
|
while (pos < end && !itt.cancel) {
|
|
|
|
samplecnt_t cur_samples = 0;
|
|
samplecnt_t const to_read = min (end - pos, block_size);
|
|
/* fill `loudest' with the loudest absolute sample at each instant, across all channels */
|
|
memset (loudest.get(), 0, sizeof (Sample) * block_size);
|
|
|
|
for (uint32_t n = 0; n < n_channels(); ++n) {
|
|
|
|
cur_samples = read_raw_internal (buf.get(), pos, to_read, n);
|
|
for (samplecnt_t i = 0; i < cur_samples; ++i) {
|
|
loudest[i] = max (loudest[i], abs (buf[i]));
|
|
}
|
|
}
|
|
|
|
/* now look for silence */
|
|
for (samplecnt_t i = 0; i < cur_samples; ++i) {
|
|
bool const silence = abs (loudest[i]) < threshold;
|
|
if (silence && !in_silence) {
|
|
/* non-silence to silence */
|
|
in_silence = true;
|
|
silence_start = pos + i + fade_length;
|
|
} else if (!silence && in_silence) {
|
|
/* silence to non-silence */
|
|
in_silence = false;
|
|
sampleoffset_t silence_end = pos + i - 1 - fade_length;
|
|
|
|
if (silence_end - silence_start >= min_length) {
|
|
silent_periods.push_back (std::make_pair (silence_start, silence_end));
|
|
}
|
|
}
|
|
}
|
|
|
|
pos += cur_samples;
|
|
itt.progress = (end - pos) / (double) length_samples();
|
|
|
|
if (cur_samples == 0) {
|
|
assert (pos >= end);
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (in_silence && !itt.cancel) {
|
|
/* last block was silent, so finish off the last period */
|
|
if (end - 1 - silence_start >= min_length + fade_length) {
|
|
silent_periods.push_back (std::make_pair (silence_start, end - 1));
|
|
}
|
|
}
|
|
|
|
itt.done = true;
|
|
|
|
return silent_periods;
|
|
}
|
|
|
|
Temporal::Range
|
|
AudioRegion::body_range () const
|
|
{
|
|
return Temporal::Range ((position() + _fade_in->back()->when).increment(), end().earlier (_fade_out->back()->when));
|
|
}
|
|
|
|
std::shared_ptr<Region>
|
|
AudioRegion::get_single_other_xfade_region (bool start) const
|
|
{
|
|
std::shared_ptr<Playlist> pl (playlist());
|
|
|
|
if (!pl) {
|
|
/* not currently in a playlist - xfade length is unbounded
|
|
(and irrelevant)
|
|
*/
|
|
return std::shared_ptr<AudioRegion> ();
|
|
}
|
|
|
|
std::shared_ptr<RegionList> rl;
|
|
|
|
if (start) {
|
|
rl = pl->regions_at (position());
|
|
} else {
|
|
rl = pl->regions_at (nt_last());
|
|
}
|
|
|
|
RegionList::iterator i;
|
|
std::shared_ptr<Region> other;
|
|
uint32_t n = 0;
|
|
|
|
/* count and find the other region in a single pass through the list */
|
|
|
|
for (i = rl->begin(); i != rl->end(); ++i) {
|
|
if ((*i).get() != this) {
|
|
other = *i;
|
|
}
|
|
++n;
|
|
}
|
|
|
|
if (n != 2) {
|
|
/* zero or multiple regions stacked here - don't care about xfades */
|
|
return std::shared_ptr<AudioRegion> ();
|
|
}
|
|
|
|
return other;
|
|
}
|
|
|
|
samplecnt_t
|
|
AudioRegion::verify_xfade_bounds (samplecnt_t len, bool start)
|
|
{
|
|
/* this is called from a UI to check on whether a new proposed
|
|
length for an xfade is legal or not. it returns the legal
|
|
length corresponding to @a len which may be shorter than or
|
|
equal to @a len itself.
|
|
*/
|
|
|
|
std::shared_ptr<Region> other = get_single_other_xfade_region (start);
|
|
samplecnt_t maxlen;
|
|
|
|
if (!other) {
|
|
/* zero or > 2 regions here, don't care about len, but
|
|
it can't be longer than the region itself.
|
|
*/
|
|
return min (length_samples(), len);
|
|
}
|
|
|
|
/* we overlap a single region. clamp the length of an xfade to
|
|
the maximum possible duration of the overlap (if the other
|
|
region were trimmed appropriately).
|
|
*/
|
|
|
|
if (start) {
|
|
maxlen = other->latest_possible_sample() - position_sample();
|
|
} else {
|
|
maxlen = last_sample() - other->earliest_possible_position().samples();
|
|
}
|
|
|
|
return min (length_samples(), min (maxlen, len));
|
|
|
|
}
|
|
|
|
bool
|
|
AudioRegion::do_export (std::string const& path) const
|
|
{
|
|
const uint32_t n_chn = n_channels ();
|
|
const samplecnt_t chunk_size = 8192;
|
|
Sample buf[chunk_size];
|
|
|
|
const int format = SF_FORMAT_FLAC | SF_FORMAT_PCM_24; // TODO preference or option
|
|
|
|
assert (!path.empty ());
|
|
assert (!Glib::file_test (path, Glib::FILE_TEST_EXISTS));
|
|
|
|
typedef std::shared_ptr<AudioGrapher::SndfileWriter<Sample>> FloatWriterPtr;
|
|
FloatWriterPtr sfw;
|
|
try {
|
|
sfw.reset (new AudioGrapher::SndfileWriter<Sample> (path, format, n_chn, audio_source ()->sample_rate (), 0));
|
|
} catch (...) {
|
|
return false;
|
|
}
|
|
|
|
AudioGrapher::Interleaver<Sample> interleaver;
|
|
interleaver.init (n_channels (), chunk_size);
|
|
interleaver.add_output (sfw);
|
|
|
|
samplecnt_t to_read = length_samples ();
|
|
samplepos_t pos = position_sample ();
|
|
samplecnt_t lsamples = _length.val().samples();
|
|
|
|
while (to_read) {
|
|
samplecnt_t this_time = min (to_read, chunk_size);
|
|
|
|
for (uint32_t chn = 0; chn < n_chn; ++chn) {
|
|
if (read_from_sources (_sources, lsamples, buf, pos, this_time, chn) != this_time) {
|
|
goto errout;
|
|
}
|
|
|
|
AudioGrapher::ConstProcessContext<Sample> context (buf, this_time, 1);
|
|
if (to_read == this_time) {
|
|
context ().set_flag (AudioGrapher::ProcessContext<Sample>::EndOfInput);
|
|
}
|
|
interleaver.input (chn)->process (context);
|
|
}
|
|
|
|
to_read -= this_time;
|
|
pos += this_time;
|
|
}
|
|
|
|
errout:
|
|
/* Drop references, close file */
|
|
interleaver.clear_outputs ();
|
|
sfw.reset ();
|
|
|
|
if (to_read != 0) {
|
|
::g_unlink (path.c_str());
|
|
}
|
|
|
|
return to_read == 0;
|
|
}
|