--- /dev/null
+/*
+ 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__ */
+
--- /dev/null
+/*
+ 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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