towards a SRC source + resampling during audition

This commit is contained in:
Robin Gareus 2014-01-18 12:31:25 +01:00
parent 96cbcf0ee4
commit 73c6122591
4 changed files with 248 additions and 2 deletions

View File

@ -52,6 +52,7 @@
#include "ardour/source_factory.h"
#include "ardour/session.h"
#include "ardour/session_directory.h"
#include "ardour/srcfilesource.h"
#include "ardour_ui.h"
#include "editing.h"
@ -389,8 +390,12 @@ SoundFileBox::audition ()
SourceFactory::createExternal (DataType::AUDIO, *_session,
path, n,
Source::Flag (0), false));
srclist.push_back(afs);
if (afs->sample_rate() != _session->nominal_frame_rate()) {
boost::shared_ptr<SrcFileSource> sfs (new SrcFileSource(*_session, afs));
srclist.push_back(sfs);
} else {
srclist.push_back(afs);
}
} catch (failed_constructor& err) {
error << _("Could not access soundfile: ") << path << endmsg;

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@ -0,0 +1,80 @@
/*
Copyright (C) 2014 Paul Davis
Written by: Robin Gareus <robin@gareus.org>
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., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __ardour_srcfilesource_h__
#define __ardour_srcfilesource_h__
#include <cstring>
#include <samplerate.h>
#include "ardour/audiofilesource.h"
#include "ardour/session.h"
namespace ARDOUR {
class SrcFileSource : public AudioFileSource {
public:
SrcFileSource (Session&, boost::shared_ptr<AudioFileSource>, SrcQuality srcq = SrcQuality(SrcQuick));
~SrcFileSource ();
int update_header (framepos_t /*when*/, struct tm&, time_t) { return 0; }
int flush_header () { return 0; }
void set_header_timeline_position () {};
void set_length (framecnt_t /*len*/) {};
float sample_rate () const { return _session.nominal_frame_rate(); }
framepos_t natural_position() const { return _source->natural_position() * _ratio;}
framecnt_t readable_length() const { return _source->readable_length() * _ratio; }
framecnt_t length (framepos_t pos) const { return _source->length(pos) * _ratio; }
bool destructive() const { return false; }
bool can_be_analysed() const { return false; }
bool clamped_at_unity() const { return false; }
protected:
framecnt_t read_unlocked (Sample *dst, framepos_t start, framecnt_t cnt) const;
framecnt_t write_unlocked (Sample */*dst*/, framecnt_t /*cnt*/) { return 0; }
int read_peaks_with_fpp (PeakData *peaks, framecnt_t npeaks, framepos_t /*start*/, framecnt_t /*cnt*/,
double /*samples_per_unit*/, framecnt_t /*fpp*/) const {
memset (peaks, 0, sizeof (PeakData) * npeaks);
return 0;
}
private:
static const uint32_t blocksize;
boost::shared_ptr<AudioFileSource> _source;
mutable SRC_STATE* _src_state;
mutable SRC_DATA _src_data;
mutable Sample* _src_buffer;
mutable framepos_t _source_position;
mutable framepos_t _target_position;
mutable double _fract_position;
double _ratio;
framecnt_t src_buffer_size;
};
} // namespace ARDOUR
#endif /* __ardour_audiofilesource_h__ */

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@ -0,0 +1,160 @@
/*
Copyright (C) 2014 Paul Davis
Written by: Robin Gareus <robin@gareus.org>
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., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "pbd/error.h"
#include "pbd/failed_constructor.h"
#include "ardour/audiofilesource.h"
#include "ardour/debug.h"
#include "ardour/srcfilesource.h"
#include "i18n.h"
using namespace ARDOUR;
using namespace PBD;
/* see disk_io_chunk_frames */
const uint32_t SrcFileSource::blocksize = 65536U;
SrcFileSource::SrcFileSource (Session& s, boost::shared_ptr<AudioFileSource> src, SrcQuality srcq)
: Source(s, DataType::AUDIO, src->name(), Flag (src->flags() & ~(Writable|Removable|RemovableIfEmpty|RemoveAtDestroy)))
, AudioFileSource (s, src->path(), Flag (src->flags() & ~(Writable|Removable|RemovableIfEmpty|RemoveAtDestroy)))
, _source (src)
, _src_state (0)
, _source_position(0)
, _target_position(0)
, _fract_position(0)
{
assert(_source->n_channels() == 1);
int src_type = SRC_SINC_BEST_QUALITY;
switch (srcq) {
case SrcBest:
src_type = SRC_SINC_BEST_QUALITY;
break;
case SrcGood:
src_type = SRC_SINC_MEDIUM_QUALITY;
break;
case SrcQuick:
src_type = SRC_SINC_FASTEST;
break;
case SrcFast:
src_type = SRC_ZERO_ORDER_HOLD;
break;
case SrcFastest:
src_type = SRC_LINEAR;
break;
}
_ratio = s.nominal_frame_rate() / _source->sample_rate();
_src_data.src_ratio = _ratio;
src_buffer_size = ceil((double)blocksize / _ratio) + 2;
_src_buffer = new float[src_buffer_size];
int err;
if ((_src_state = src_new (src_type, 1, &err)) == 0) {
error << string_compose(_("Import: src_new() failed : %1"), src_strerror (err)) << endmsg ;
throw failed_constructor ();
}
}
SrcFileSource::~SrcFileSource ()
{
DEBUG_TRACE (DEBUG::AudioPlayback, "SrcFileSource::~SrcFileSource\n");
_src_state = src_delete (_src_state) ;
delete [] _src_buffer;
}
framecnt_t
SrcFileSource::read_unlocked (Sample *dst, framepos_t start, framecnt_t cnt) const
{
int err;
const double srccnt = cnt / _ratio;
if (_target_position != start) {
DEBUG_TRACE (DEBUG::AudioPlayback, string_compose ("SRC: reset %1 -> %2\n", _target_position, start));
src_reset(_src_state);
_fract_position = 0;
_source_position = start / _ratio;
_target_position = start;
}
const framecnt_t scnt = ceilf(srccnt - _fract_position);
_fract_position += (scnt - srccnt);
#ifndef NDEBUG
if (scnt >= src_buffer_size) {
DEBUG_TRACE (DEBUG::AudioPlayback, string_compose ("SRC: CRASH AHEAD :) %1 >= %2 (fract=%3, cnt=%4)\n",
scnt, src_buffer_size, _fract_position, cnt));
}
#endif
assert(scnt < src_buffer_size);
_src_data.input_frames = _source->read (_src_buffer, _source_position, scnt);
if ((framecnt_t) _src_data.input_frames < scnt
|| _source_position + scnt >= _source->length(0)) {
_src_data.end_of_input = true;
_target_position += _src_data.input_frames * _ratio;
DEBUG_TRACE (DEBUG::AudioPlayback, "SRC: END OF INPUT\n");
} else {
_src_data.end_of_input = false;
_target_position += cnt;
}
_src_data.output_frames = cnt;
_src_data.data_in = _src_buffer;
_src_data.data_out = dst;
if (_src_data.end_of_input) {
_src_data.output_frames = std::min ((long)floor(_src_data.input_frames * _ratio), _src_data.output_frames);
}
if ((err = src_process (_src_state, &_src_data))) {
error << string_compose(_("SrcFileSource: %1"), src_strerror (err)) << endmsg ;
return 0;
}
if (_src_data.end_of_input && _src_data.output_frames_gen <= 0) {
return 0;
}
_source_position += _src_data.input_frames_used;
framepos_t saved_target = _target_position;
framecnt_t generated = _src_data.output_frames_gen;
while (generated < cnt) {
DEBUG_TRACE (DEBUG::AudioPlayback, string_compose ("SRC: recurse for %1 samples\n", cnt - generated));
framecnt_t g = read_unlocked(dst + generated, _target_position, cnt - generated);
generated += g;
if (g == 0) break;
}
_target_position = saved_target;
DEBUG_TRACE (DEBUG::AudioPlayback, string_compose ("SRC: in: %1-> want: %2 || got: %3 total: %4\n",
_src_data.input_frames, _src_data.output_frames, _src_data.output_frames_gen, generated));
return generated;
}

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@ -199,6 +199,7 @@ libardour_sources = [
'source.cc',
'source_factory.cc',
'speakers.cc',
'srcfilesource.cc',
'strip_silence.cc',
'revision.cc',
'tape_file_matcher.cc',