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ardour/libs/ardour/audioregion.cc

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/*
* Copyright (C) 2006-2014 David Robillard <d@drobilla.net>
* Copyright (C) 2006-2017 Paul Davis <paul@linuxaudiosystems.com>
* Copyright (C) 2007-2012 Carl Hetherington <carl@carlh.net>
* Copyright (C) 2012-2019 Robin Gareus <robin@gareus.org>
* Copyright (C) 2015-2018 Ben Loftis <ben@harrisonconsoles.com>
* Copyright (C) 2016-2017 Nick Mainsbridge <mainsbridge@gmail.com>
* Copyright (C) 2016 Tim Mayberry <mojofunk@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <algorithm>
#include <cfloat>
#include <climits>
#include <cmath>
#include <memory>
#include <set>
#include <boost/scoped_array.hpp>
#include <glibmm/fileutils.h>
#include <glibmm/threads.h>
#include "pbd/gstdio_compat.h"
#include "pbd/basename.h"
#include "pbd/xml++.h"
#include "pbd/enumwriter.h"
#include "pbd/convert.h"
#include "pbd/progress.h"
#include "evoral/Curve.h"
#include "ardour/audioengine.h"
#include "ardour/analysis_graph.h"
#include "ardour/audioregion.h"
#include "ardour/buffer_manager.h"
#include "ardour/session.h"
#include "ardour/dB.h"
#include "ardour/debug.h"
#include "ardour/event_type_map.h"
#include "ardour/playlist.h"
#include "ardour/audiofilesource.h"
#include "ardour/region_factory.h"
#include "ardour/region_fx_plugin.h"
#include "ardour/runtime_functions.h"
#include "ardour/sndfilesource.h"
#include "ardour/transient_detector.h"
#include "ardour/parameter_descriptor.h"
#include "audiographer/general/interleaver.h"
#include "audiographer/general/sample_format_converter.h"
#include "audiographer/sndfile/sndfile_writer.h"
#include "pbd/i18n.h"
#include <locale.h>
using namespace std;
using namespace ARDOUR;
using namespace PBD;
#define S2SC(s) Temporal::samples_to_superclock (s, TEMPORAL_SAMPLE_RATE)
#define SC2S(s) Temporal::superclock_to_samples (s, TEMPORAL_SAMPLE_RATE)
namespace ARDOUR {
namespace Properties {
PBD::PropertyDescriptor<bool> envelope_active;
PBD::PropertyDescriptor<bool> default_fade_in;
PBD::PropertyDescriptor<bool> default_fade_out;
PBD::PropertyDescriptor<bool> fade_in_active;
PBD::PropertyDescriptor<bool> fade_out_active;
PBD::PropertyDescriptor<float> scale_amplitude;
PBD::PropertyDescriptor<std::shared_ptr<AutomationList> > fade_in;
PBD::PropertyDescriptor<std::shared_ptr<AutomationList> > inverse_fade_in;
PBD::PropertyDescriptor<std::shared_ptr<AutomationList> > fade_out;
PBD::PropertyDescriptor<std::shared_ptr<AutomationList> > inverse_fade_out;
PBD::PropertyDescriptor<std::shared_ptr<AutomationList> > envelope;
}
}
/* Curve manipulations */
static void
reverse_curve (std::shared_ptr<Evoral::ControlList> dst, std::shared_ptr<const Evoral::ControlList> src)
{
const timepos_t end = src->when(false);
// TODO read-lock of src (!)
for (Evoral::ControlList::const_reverse_iterator it = src->rbegin(); it!=src->rend(); it++) {
/* ugh ... the double "distance" calls (with totally different
semantics ... horrible
*/
dst->fast_simple_add (timepos_t ((*it)->when.distance (end)), (*it)->value);
}
}
static void
generate_inverse_power_curve (std::shared_ptr<Evoral::ControlList> dst, std::shared_ptr<const Evoral::ControlList> src)
{
// calc inverse curve using sum of squares
for (Evoral::ControlList::const_iterator it = src->begin(); it!=src->end(); ++it ) {
float value = (*it)->value;
value = 1 - powf(value,2);
value = sqrtf(value);
dst->fast_simple_add ((*it)->when, value );
}
}
static void
generate_db_fade (std::shared_ptr<Evoral::ControlList> dst, double len, int num_steps, float dB_drop)
{
dst->clear ();
dst->fast_simple_add (timepos_t (Temporal::AudioTime), 1);
//generate a fade-out curve by successively applying a gain drop
float fade_speed = dB_to_coefficient(dB_drop / (float) num_steps);
float coeff = GAIN_COEFF_UNITY;
for (int i = 1; i < (num_steps-1); i++) {
coeff *= fade_speed;
dst->fast_simple_add (timepos_t (samplepos_t (len*(double)i/(double)num_steps)), coeff);
}
dst->fast_simple_add (timepos_t ((samplepos_t)len), GAIN_COEFF_SMALL);
}
static void
merge_curves (std::shared_ptr<Evoral::ControlList> dst,
std::shared_ptr<const Evoral::ControlList> curve1,
std::shared_ptr<const Evoral::ControlList> curve2)
{
Evoral::ControlList::EventList::size_type size = curve1->size();
//curve lengths must match for now
if (size != curve2->size()) {
return;
}
Evoral::ControlList::const_iterator c1 = curve1->begin();
int count = 0;
for (Evoral::ControlList::const_iterator c2 = curve2->begin(); c2!=curve2->end(); c2++ ) {
float v1 = accurate_coefficient_to_dB((*c1)->value);
float v2 = accurate_coefficient_to_dB((*c2)->value);
double interp = v1 * ( 1.0-( (double)count / (double)size) );
interp += v2 * ( (double)count / (double)size );
interp = dB_to_coefficient(interp);
dst->fast_simple_add ((*c1)->when, interp );
c1++;
count++;
}
}
void
AudioRegion::make_property_quarks ()
{
Properties::envelope_active.property_id = g_quark_from_static_string (X_("envelope-active"));
DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for envelope-active = %1\n", Properties::envelope_active.property_id));
Properties::default_fade_in.property_id = g_quark_from_static_string (X_("default-fade-in"));
DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for default-fade-in = %1\n", Properties::default_fade_in.property_id));
Properties::default_fade_out.property_id = g_quark_from_static_string (X_("default-fade-out"));
DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for default-fade-out = %1\n", Properties::default_fade_out.property_id));
Properties::fade_in_active.property_id = g_quark_from_static_string (X_("fade-in-active"));
DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for fade-in-active = %1\n", Properties::fade_in_active.property_id));
Properties::fade_out_active.property_id = g_quark_from_static_string (X_("fade-out-active"));
DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for fade-out-active = %1\n", Properties::fade_out_active.property_id));
Properties::scale_amplitude.property_id = g_quark_from_static_string (X_("scale-amplitude"));
DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for scale-amplitude = %1\n", Properties::scale_amplitude.property_id));
Properties::fade_in.property_id = g_quark_from_static_string (X_("FadeIn"));
DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for FadeIn = %1\n", Properties::fade_in.property_id));
Properties::inverse_fade_in.property_id = g_quark_from_static_string (X_("InverseFadeIn"));
DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for InverseFadeIn = %1\n", Properties::inverse_fade_in.property_id));
Properties::fade_out.property_id = g_quark_from_static_string (X_("FadeOut"));
DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for FadeOut = %1\n", Properties::fade_out.property_id));
Properties::inverse_fade_out.property_id = g_quark_from_static_string (X_("InverseFadeOut"));
DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for InverseFadeOut = %1\n", Properties::inverse_fade_out.property_id));
Properties::envelope.property_id = g_quark_from_static_string (X_("Envelope"));
DEBUG_TRACE (DEBUG::Properties, string_compose ("quark for Envelope = %1\n", Properties::envelope.property_id));
}
void
AudioRegion::register_properties ()
{
/* no need to register parent class properties */
add_property (_envelope_active);
add_property (_default_fade_in);
add_property (_default_fade_out);
add_property (_fade_in_active);
add_property (_fade_out_active);
add_property (_scale_amplitude);
add_property (_fade_in);
add_property (_inverse_fade_in);
add_property (_fade_out);
add_property (_inverse_fade_out);
add_property (_envelope);
}
#define AUDIOREGION_STATE_DEFAULT(tdp) \
_envelope_active (Properties::envelope_active, false) \
, _default_fade_in (Properties::default_fade_in, true) \
, _default_fade_out (Properties::default_fade_out, true) \
, _fade_in_active (Properties::fade_in_active, true) \
, _fade_out_active (Properties::fade_out_active, true) \
, _scale_amplitude (Properties::scale_amplitude, 1.0) \
, _fade_in (Properties::fade_in, std::shared_ptr<AutomationList> (new AutomationList (Evoral::Parameter (FadeInAutomation), tdp))) \
, _inverse_fade_in (Properties::inverse_fade_in, std::shared_ptr<AutomationList> (new AutomationList (Evoral::Parameter (FadeInAutomation), tdp))) \
, _fade_out (Properties::fade_out, std::shared_ptr<AutomationList> (new AutomationList (Evoral::Parameter (FadeOutAutomation), tdp))) \
, _inverse_fade_out (Properties::inverse_fade_out, std::shared_ptr<AutomationList> (new AutomationList (Evoral::Parameter (FadeOutAutomation), tdp)))
#define AUDIOREGION_COPY_STATE(other) \
_envelope_active (Properties::envelope_active, other->_envelope_active) \
, _default_fade_in (Properties::default_fade_in, other->_default_fade_in) \
, _default_fade_out (Properties::default_fade_out, other->_default_fade_out) \
, _fade_in_active (Properties::fade_in_active, other->_fade_in_active) \
, _fade_out_active (Properties::fade_out_active, other->_fade_out_active) \
, _scale_amplitude (Properties::scale_amplitude, other->_scale_amplitude) \
, _fade_in (Properties::fade_in, std::shared_ptr<AutomationList> (new AutomationList (*other->_fade_in.val()))) \
, _inverse_fade_in (Properties::fade_in, std::shared_ptr<AutomationList> (new AutomationList (*other->_inverse_fade_in.val()))) \
, _fade_out (Properties::fade_in, std::shared_ptr<AutomationList> (new AutomationList (*other->_fade_out.val()))) \
, _inverse_fade_out (Properties::fade_in, std::shared_ptr<AutomationList> (new AutomationList (*other->_inverse_fade_out.val())))
/* a Session will reset these to its chosen defaults by calling AudioRegion::set_default_fade() */
void
AudioRegion::init ()
{
register_properties ();
suspend_property_changes();
set_default_fades ();
set_default_envelope ();
resume_property_changes();
listen_to_my_curves ();
connect_to_analysis_changed ();
connect_to_header_position_offset_changed ();
_fx_pos = _cache_start = _cache_end = -1;
_fx_block_size = 0;
_fx_latent_read = false;
}
void
AudioRegion::send_change (const PropertyChange& what_changed)
{
PropertyChange our_interests;
our_interests.add (Properties::fade_in_active);
our_interests.add (Properties::fade_out_active);
our_interests.add (Properties::scale_amplitude);
our_interests.add (Properties::envelope_active);
our_interests.add (Properties::envelope);
our_interests.add (Properties::fade_in);
our_interests.add (Properties::fade_out);
if (what_changed.contains (our_interests)) {
_invalidated.exchange (true);
}
Region::send_change (what_changed);
}
void
AudioRegion::copy_plugin_state (std::shared_ptr<const AudioRegion> other)
{
/* state cannot copied in Region, because when running Region's c'tor
* the AudioRegion does not yet exist, and virtual _add_plugin
* of the parent class is called
*/
Glib::Threads::RWLock::ReaderLock lm (other->_fx_lock);
for (auto const& i : other->_plugins) {
XMLNode& state = i->get_state ();
state.remove_property ("count");
PBD::Stateful::ForceIDRegeneration force_ids;
std::shared_ptr<RegionFxPlugin> rfx (new RegionFxPlugin (_session, Temporal::AudioTime));
rfx->set_state (state, Stateful::current_state_version);
if (!_add_plugin (rfx, std::shared_ptr<RegionFxPlugin>(), true)) {
continue;
}
_plugins.push_back (rfx);
delete &state;
}
fx_latency_changed (true);
}
/** Constructor for use by derived types only */
AudioRegion::AudioRegion (Session& s, timepos_t const & start, timecnt_t const & len, std::string name)
: Region (s, start, len, name, DataType::AUDIO)
, AUDIOREGION_STATE_DEFAULT(Temporal::TimeDomainProvider (Temporal::AudioTime))
, _envelope (Properties::envelope, std::shared_ptr<AutomationList> (new AutomationList (Evoral::Parameter(EnvelopeAutomation), Temporal::TimeDomainProvider (Temporal::AudioTime))))
, _automatable (s, Temporal::TimeDomainProvider (Temporal::AudioTime))
, _fade_in_suspended (0)
, _fade_out_suspended (0)
{
init ();
assert (_sources.size() == _master_sources.size());
}
/** Basic AudioRegion constructor */
AudioRegion::AudioRegion (const SourceList& srcs)
: Region (srcs)
, AUDIOREGION_STATE_DEFAULT(Temporal::TimeDomainProvider (Temporal::AudioTime))
, _envelope (Properties::envelope, std::shared_ptr<AutomationList> (new AutomationList (Evoral::Parameter(EnvelopeAutomation), Temporal::TimeDomainProvider (Temporal::AudioTime))))
, _automatable(srcs[0]->session(), Temporal::TimeDomainProvider (Temporal::AudioTime))
, _fade_in_suspended (0)
, _fade_out_suspended (0)
{
init ();
assert (_sources.size() == _master_sources.size());
}
AudioRegion::AudioRegion (std::shared_ptr<const AudioRegion> other)
: Region (other)
, AUDIOREGION_COPY_STATE (other)
/* As far as I can see, the _envelope's times are relative to region position, and have nothing
* to do with sources (and hence _start). So when we copy the envelope, we just use the supplied offset.
*/
, _envelope (Properties::envelope, std::shared_ptr<AutomationList> (new AutomationList (*other->_envelope.val(), timepos_t (Temporal::AudioTime), other->len_as_tpos ())))
, _automatable (other->session(), Temporal::TimeDomainProvider (Temporal::AudioTime))
, _fade_in_suspended (0)
, _fade_out_suspended (0)
{
/* don't use init here, because we got fade in/out from the other region
*/
register_properties ();
listen_to_my_curves ();
connect_to_analysis_changed ();
connect_to_header_position_offset_changed ();
_fx_pos = _cache_start = _cache_end = -1;
_fx_block_size = 0;
_fx_latent_read = false;
copy_plugin_state (other);
assert(_type == DataType::AUDIO);
assert (_sources.size() == _master_sources.size());
}
AudioRegion::AudioRegion (std::shared_ptr<const AudioRegion> other, timecnt_t const & offset)
: Region (other, offset)
, AUDIOREGION_COPY_STATE (other)
/* As far as I can see, the _envelope's times are relative to region position, and have nothing
to do with sources (and hence _start). So when we copy the envelope, we just use the supplied offset.
*/
, _envelope (Properties::envelope, std::shared_ptr<AutomationList> (new AutomationList (*other->_envelope.val(), timepos_t (offset.samples()), other->len_as_tpos ())))
, _automatable (other->session(), Temporal::TimeDomainProvider (Temporal::AudioTime))
, _fade_in_suspended (0)
, _fade_out_suspended (0)
{
/* don't use init here, because we got fade in/out from the other region
*/
register_properties ();
listen_to_my_curves ();
connect_to_analysis_changed ();
connect_to_header_position_offset_changed ();
_fx_pos = _cache_start = _cache_end = -1;
_fx_block_size = 0;
_fx_latent_read = false;
copy_plugin_state (other);
assert(_type == DataType::AUDIO);
assert (_sources.size() == _master_sources.size());
}
AudioRegion::AudioRegion (std::shared_ptr<const AudioRegion> other, const SourceList& srcs)
: Region (std::static_pointer_cast<const Region>(other), srcs)
, AUDIOREGION_COPY_STATE (other)
, _envelope (Properties::envelope, std::shared_ptr<AutomationList> (new AutomationList (*other->_envelope.val())))
, _automatable (other->session(), Temporal::TimeDomainProvider (Temporal::AudioTime))
, _fade_in_suspended (0)
, _fade_out_suspended (0)
{
/* make-a-sort-of-copy-with-different-sources constructor (used by audio filter) */
register_properties ();
listen_to_my_curves ();
connect_to_analysis_changed ();
connect_to_header_position_offset_changed ();
_fx_pos = _cache_start = _cache_end = -1;
_fx_block_size = 0;
_fx_latent_read = false;
copy_plugin_state (other);
assert (_sources.size() == _master_sources.size());
}
AudioRegion::AudioRegion (SourceList& srcs)
: Region (srcs)
, AUDIOREGION_STATE_DEFAULT(srcs[0]->session())
, _envelope (Properties::envelope, std::shared_ptr<AutomationList> (new AutomationList(Evoral::Parameter(EnvelopeAutomation), Temporal::TimeDomainProvider (Temporal::AudioTime))))
, _automatable(srcs[0]->session(), Temporal::TimeDomainProvider (Temporal::AudioTime))
, _fade_in_suspended (0)
, _fade_out_suspended (0)
{
init ();
assert(_type == DataType::AUDIO);
assert (_sources.size() == _master_sources.size());
}
AudioRegion::~AudioRegion ()
{
for (auto const& rfx : _plugins) {
rfx->drop_references ();
}
}
void
AudioRegion::post_set (const PropertyChange& /*ignored*/)
{
if (!_sync_marked) {
_sync_position = _start;
}
/* return to default fades if the existing ones are too long */
if (_left_of_split) {
if (_fade_in->when(false) >= len_as_tpos ()) {
set_default_fade_in ();
}
set_default_fade_out ();
_left_of_split = false;
}
if (_right_of_split) {
if (_fade_out->when(false) >= len_as_tpos ()) {
set_default_fade_out ();
}
set_default_fade_in ();
_right_of_split = false;
}
/* If _length changed, adjust our gain envelope accordingly */
_envelope->truncate_end (len_as_tpos ());
}
void
AudioRegion::connect_to_analysis_changed ()
{
for (SourceList::const_iterator i = _sources.begin(); i != _sources.end(); ++i) {
(*i)->AnalysisChanged.connect_same_thread (*this, boost::bind (&AudioRegion::maybe_invalidate_transients, this));
}
}
void
AudioRegion::connect_to_header_position_offset_changed ()
{
set<std::shared_ptr<Source> > unique_srcs;
for (SourceList::const_iterator i = _sources.begin(); i != _sources.end(); ++i) {
/* connect only once to HeaderPositionOffsetChanged, even if sources are replicated
*/
if (unique_srcs.find (*i) == unique_srcs.end ()) {
unique_srcs.insert (*i);
std::shared_ptr<AudioFileSource> afs = std::dynamic_pointer_cast<AudioFileSource> (*i);
if (afs) {
afs->HeaderPositionOffsetChanged.connect_same_thread (*this, boost::bind (&AudioRegion::source_offset_changed, this));
}
}
}
}
void
AudioRegion::listen_to_my_curves ()
{
_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, 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,
gain_t* 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);
uint32_t const n_chn = n_channels ();
if (n_chn == 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 */
}
const samplecnt_t esamples = lsamples - internal_offset;
assert (esamples >= 0);
if ((to_read = min (cnt, esamples)) == 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;
}
}
Glib::Threads::Mutex::Lock cl (_cache_lock);
if (chan_n == 0 && _invalidated.exchange (false)) {
_cache_start = _cache_end = -1;
}
boost::scoped_array<gain_t> gain_array;
boost::scoped_array<Sample> mixdown_array;
// TODO optimize mono reader, w/o plugins -> old code
if (n_chn > 1 && _cache_start < _cache_end && internal_offset >= _cache_start && internal_offset + to_read <= _cache_end) {
DEBUG_TRACE (DEBUG::AudioPlayback, string_compose ("Region '%1' channel: %2 copy from cache %3 - %4 to_read: %5\n",
name(), chan_n, internal_offset, internal_offset + to_read, to_read));
copy_vector (mixdown_buffer, _readcache.get_audio (chan_n).data (internal_offset - _cache_start), to_read);
cl.release ();
} else {
Glib::Threads::RWLock::ReaderLock lm (_fx_lock);
bool have_fx = !_plugins.empty ();
uint32_t fx_latency = _fx_latency;
lm.release ();
ChanCount cc (DataType::AUDIO, n_channels ());
_readcache.ensure_buffers (cc, to_read + _fx_latency);
samplecnt_t n_read = to_read; //< data to read from disk
samplecnt_t n_proc = to_read; //< silence pad data to process
samplepos_t readat = pos;
sampleoffset_t offset = internal_offset;
//printf ("READ Cache end %ld pos %ld\n", _cache_end, readat);
if (_cache_end != readat && fx_latency > 0) {
_fx_latent_read = true;
n_proc += fx_latency;
n_read = min (to_read + fx_latency, esamples);
mixdown_array.reset (new Sample[n_proc]);
mixdown_buffer = mixdown_array.get ();
gain_array.reset (new gain_t[n_proc]);
gain_buffer = gain_array.get ();
}
if (!_fx_latent_read && fx_latency > 0) {
offset += fx_latency;
readat += fx_latency;
n_read = max<samplecnt_t> (0, min (to_read, lsamples - offset));
}
DEBUG_TRACE (DEBUG::AudioPlayback, string_compose ("Region '%1' channel: %2 read: %3 - %4 (%5) to_read: %6 offset: %7 with fx: %8 fx_latency: %9\n",
name(), chan_n, readat, readat + n_read, n_read, to_read, internal_offset, have_fx, fx_latency));
_readcache.ensure_buffers (cc, n_proc);
if (n_read < n_proc) {
//printf ("SILENCE PAD rd: %ld proc: %ld\n", n_read, n_proc);
/* silence pad, process tail of latent effects */
memset (&mixdown_buffer[n_read], 0, sizeof (Sample)* (n_proc - n_read));
_readcache.silence (n_proc - n_read, n_read);
}
/* reset in case read fails we return early */
_cache_start = _cache_end = -1;
for (uint32_t chn = 0; chn < n_chn; ++chn) {
/* 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, readat, n_read, chn) != n_read) {
return 0; // XXX
}
/* APPLY REGULAR GAIN CURVES AND SCALING TO mixdown_buffer */
if (envelope_active()) {
_envelope->curve().get_vector (timepos_t (offset), timepos_t (offset + n_read), gain_buffer, n_read);
if (_scale_amplitude != 1.0f) {
for (samplecnt_t n = 0; n < n_read; ++n) {
mixdown_buffer[n] *= gain_buffer[n] * _scale_amplitude;
}
} else {
for (samplecnt_t n = 0; n < n_read; ++n) {
mixdown_buffer[n] *= gain_buffer[n];
}
}
} else if (_scale_amplitude != 1.0f) {
apply_gain_to_buffer (mixdown_buffer, n_read, _scale_amplitude);
}
/* for mono regions no cache is required, unless there are
* regionFX, which use the _readcache BufferSet.
*/
if (n_chn > 1 || have_fx) {
_readcache.get_audio (chn).read_from (mixdown_buffer, n_proc);
}
}
/* apply region FX to all channels */
if (have_fx) {
const_cast<AudioRegion*>(this)->apply_region_fx (_readcache, offset, offset + n_proc, n_proc);
}
/* for mono regions without plugins, mixdown_buffer is valid as-is */
if (n_chn > 1 || have_fx) {
/* copy data for current channel */
copy_vector (mixdown_buffer, _readcache.get_audio (chan_n).data (), to_read);
}
_cache_start = internal_offset;
_cache_end = internal_offset + to_read;
cl.release ();
}
/* 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));
copy_vector (buf + fade_in_limit, mixdown_buffer + fade_in_limit, N);
} 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::TimeDomainProvider (Temporal::AudioTime)));
std::shared_ptr<Evoral::ControlList> c2 (new Evoral::ControlList (FadeInAutomation, desc, Temporal::TimeDomainProvider (Temporal::AudioTime)));
std::shared_ptr<Evoral::ControlList> c3 (new Evoral::ControlList (FadeInAutomation, desc, Temporal::TimeDomainProvider (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::TimeDomainProvider (Temporal::AudioTime)));
std::shared_ptr<Evoral::ControlList> c2 (new Evoral::ControlList (FadeOutAutomation, desc, Temporal::TimeDomainProvider (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.
*/
timepos_t tend (len_as_tpos ());
_envelope->freeze ();
_envelope->truncate_end (tend);
_envelope->thaw ();
foreach_plugin ([tend](std::weak_ptr<RegionFxPlugin> wfx)
{
shared_ptr<RegionFxPlugin> rfx = wfx.lock ();
if (rfx) {
rfx->truncate_automation_end (tend);
}
});
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 */
timecnt_t tas (timecnt_t::from_samples (length().samples ()));
_envelope->truncate_start (tas);
foreach_plugin ([tas](std::weak_ptr<RegionFxPlugin> wfx)
{
shared_ptr<RegionFxPlugin> rfx = wfx.lock ();
if (rfx) {
rfx->truncate_automation_start (tas);
}
});
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 > Metering, 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;
}
bool
AudioRegion::_add_plugin (std::shared_ptr<RegionFxPlugin> rfx, std::shared_ptr<RegionFxPlugin> before, bool from_set_state)
{
ChanCount in (DataType::AUDIO, n_channels ());
ChanCount out (in);
if (!rfx->can_support_io_configuration (in, out)) {
return false;
}
if (in.n_audio () > out.n_audio ()) {
return false;
}
if (!rfx->configure_io (in, out)) {
return false;
}
ChanCount fx_cc;
{
Glib::Threads::RWLock::ReaderLock lm (_fx_lock, Glib::Threads::NOT_LOCK);
if (!from_set_state) {
lm.acquire();
}
ChanCount cc (DataType::AUDIO, n_channels ());
fx_cc = ChanCount::max (in, out);
fx_cc = ChanCount::max (fx_cc, rfx->required_buffers ());
for (auto const& i : _plugins) {
fx_cc = ChanCount::max (fx_cc, i->required_buffers ());
}
}
DEBUG_TRACE (DEBUG::RegionFx, string_compose ("Audio Region Fx required ChanCount: %1\n", fx_cc));
_session.ensure_buffers_unlocked (fx_cc);
/* subscribe to parameter changes */
ControllableSet acs;
rfx->automatables (acs);
for (auto& ec : acs) {
std::shared_ptr<AutomationControl> ac (std::dynamic_pointer_cast<AutomationControl>(ec));
std::weak_ptr<AutomationControl> wc (ac);
ec->Changed.connect_same_thread (*this, [this, wc] (bool, PBD::Controllable::GroupControlDisposition)
{
std::shared_ptr<AutomationControl> ac = wc.lock ();
if (ac && ac->automation_playback ()) {
return;
}
if (!_invalidated.exchange (true)) {
send_change (PropertyChange (Properties::region_fx)); // trigger DiskReader overwrite
}
});
if (!ac->alist ()) {
continue;
}
ac->alist()->StateChanged.connect_same_thread (*this, [this] ()
{
if (!_invalidated.exchange (true)) {
send_change (PropertyChange (Properties::region_fx)); // trigger DiskReader overwrite
}
});
}
rfx->LatencyChanged.connect_same_thread (*this, boost::bind (&AudioRegion::fx_latency_changed, this, false));
rfx->set_block_size (_session.get_block_size ());
if (from_set_state) {
return true;
}
{
Glib::Threads::RWLock::WriterLock lm (_fx_lock);
RegionFxList::iterator loc = _plugins.end ();
if (before) {
loc = find (_plugins.begin (), _plugins.end (), before);
}
_plugins.insert (loc, rfx);
}
rfx->set_default_automation (len_as_tpos ());
fx_latency_changed (true);
if (!_invalidated.exchange (true)) {
send_change (PropertyChange (Properties::region_fx)); // trigger DiskReader overwrite
}
RegionFxChanged (); /* EMIT SIGNAL */
return true;
}
bool
AudioRegion::remove_plugin (std::shared_ptr<RegionFxPlugin> fx)
{
Glib::Threads::RWLock::WriterLock lm (_fx_lock);
auto i = find (_plugins.begin(), _plugins.end(), fx);
if (i == _plugins.end ()) {
return false;
}
_plugins.erase (i);
lm.release ();
fx->drop_references ();
fx_latency_changed (true);
if (!_invalidated.exchange (true)) {
send_change (PropertyChange (Properties::region_fx)); // trigger DiskReader overwrite
}
RegionFxChanged (); /* EMIT SIGNAL */
return true;
}
void
AudioRegion::reorder_plugins (RegionFxList const& new_order)
{
Region::reorder_plugins (new_order);
if (!_invalidated.exchange (true)) {
send_change (PropertyChange (Properties::region_fx)); // trigger DiskReader overwrite
}
RegionFxChanged (); /* EMIT SIGNAL */
}
void
AudioRegion::fx_latency_changed (bool no_emit)
{
uint32_t l = 0;
for (auto const& rfx : _plugins) {
l += rfx->effective_latency ();
}
if (l == _fx_latency) {
return;
}
_fx_latency = l;
if (no_emit) {
return;
}
if (!_invalidated.exchange (true)) {
send_change (PropertyChange (Properties::region_fx)); // trigger DiskReader overwrite
}
}
void
AudioRegion::apply_region_fx (BufferSet& bufs, samplepos_t start_sample, samplepos_t end_sample, samplecnt_t n_samples)
{
Glib::Threads::RWLock::ReaderLock lm (_fx_lock);
if (_plugins.empty ()) {
return;
}
pframes_t block_size = _session.get_block_size ();
if (_fx_block_size != block_size) {
_fx_block_size = block_size;
for (auto const& rfx : _plugins) {
rfx->set_block_size (_session.get_block_size ());
}
}
samplecnt_t latency_offset = 0;
for (auto const& rfx : _plugins) {
if (_fx_pos != start_sample) {
rfx->flush ();
}
samplecnt_t remain = n_samples;
samplecnt_t offset = 0;
samplecnt_t latency = rfx->effective_latency ();
while (remain > 0) {
pframes_t run = std::min <pframes_t> (remain, block_size);
if (!rfx->run (bufs, start_sample + offset - latency_offset, end_sample + offset - latency_offset, position().samples(), run, offset)) {
lm.release ();
/* this triggers a re-read */
const_cast<AudioRegion*>(this)->remove_plugin (rfx);
return;
}
remain -= run;
offset += run;
}
if (_fx_latent_read && latency > 0) {
for (uint32_t c = 0; c < n_channels (); ++c) {
Sample* to = _readcache.get_audio (c).data();
Sample* from = _readcache.get_audio (c).data(latency);
// XXX can left to right copy_vector() work here?
memmove (to, from, (n_samples - latency) * sizeof(Sample));
}
n_samples -= latency;
}
if (!_fx_latent_read) {
latency_offset += latency;
}
}
_fx_pos = end_sample;
_fx_latent_read = false;
}
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