fix up BWF creation; make CoreAudioSource work; add CAImportable; refactor SourceFactory to try sndfile first, CoreAudio second

git-svn-id: svn://localhost/ardour2/branches/2.0-ongoing@3126 d708f5d6-7413-0410-9779-e7cbd77b26cf

gtk2_ardour/editor.h

@@ -1075,7 +1075,10 @@ class Editor : public PublicEditor
 	SoundFileOmega* sfbrowser;
 	
 	void bring_in_external_audio (Editing::ImportMode mode,  nframes64_t& pos);
+
+	void _do_import (vector<Glib::ustring> paths, Editing::ImportDisposition, Editing::ImportMode mode, ARDOUR::SrcQuality, nframes64_t&);
 	void do_import (vector<Glib::ustring> paths, Editing::ImportDisposition, Editing::ImportMode mode, ARDOUR::SrcQuality, nframes64_t&);
+	bool idle_do_import (vector<Glib::ustring> paths, Editing::ImportDisposition, Editing::ImportMode mode, ARDOUR::SrcQuality, nframes64_t&);
 
 	void _do_embed (vector<Glib::ustring> paths, Editing::ImportDisposition, Editing::ImportMode mode,  nframes64_t&);
 	void do_embed (vector<Glib::ustring> paths, Editing::ImportDisposition, Editing::ImportMode mode,  nframes64_t&);

gtk2_ardour/editor_audio_import.cc

@@ -280,8 +280,25 @@ Editor::get_nth_selected_audio_track (int nth) const
 	return atv->audio_track();
 }	
 
+bool
+Editor::idle_do_import (vector<ustring> paths, ImportDisposition chns, ImportMode mode, SrcQuality quality, nframes64_t& pos)
+{
+	_do_import (paths, chns, mode, quality, pos);
+	return false;
+}
+
 void
 Editor::do_import (vector<ustring> paths, ImportDisposition chns, ImportMode mode, SrcQuality quality, nframes64_t& pos)
+{
+#ifdef GTKOSX
+	Glib::signal_idle().connect (bind (mem_fun (*this, &Editor::idle_do_import), paths, chns, mode, quality, pos));
+#else
+	_do_import (paths, chns, mode, quality, pos);
+#endif
+}
+
+void
+Editor::_do_import (vector<ustring> paths, ImportDisposition chns, ImportMode mode, SrcQuality quality, nframes64_t& pos)
 {
 	boost::shared_ptr<AudioTrack> track;
 	vector<ustring> to_import;
@@ -653,6 +670,9 @@ Editor::embed_sndfiles (vector<Glib::ustring> paths, bool multifile,
 				boost::shared_ptr<Source> s;
 
 				if ((s = session->source_by_path_and_channel (path, n)) == 0) {
+
+					cerr << "add embed/import source with defer_peaks = true\n";
+
 					source = boost::dynamic_pointer_cast<AudioFileSource> (SourceFactory::createReadable 
 											       (*session, path,  n,
 												(mode == ImportAsTapeTrack ? 

gtk2_ardour/editor_canvas.cc

@@ -498,7 +498,11 @@ Editor::drop_paths (const RefPtr<Gdk::DragContext>& context,
 
 		frame = 0;
 
-		do_embed (paths, Editing::ImportDistinctFiles, ImportAsTrack, frame);
+		if (Profile->get_sae() || Config->get_only_copy_imported_files()) {
+			do_import (paths, Editing::ImportDistinctFiles, Editing::ImportAsTrack, SrcBest, frame); 
+		} else {
+			do_embed (paths, Editing::ImportDistinctFiles, ImportAsTrack, frame);
+		}
 		
 	} else if ((tv = dynamic_cast<AudioTimeAxisView*>(tvp)) != 0) {
 
@@ -507,7 +511,12 @@ Editor::drop_paths (const RefPtr<Gdk::DragContext>& context,
 		if (tv->get_diskstream()) {
 			/* select the track, then embed */
 			selection->set (tv);
-			do_embed (paths, Editing::ImportDistinctFiles, ImportToTrack, frame);
+
+			if (Profile->get_sae() || Config->get_only_copy_imported_files()) {
+				do_import (paths, Editing::ImportDistinctFiles, Editing::ImportToTrack, SrcBest, frame); 
+			} else {
+				do_embed (paths, Editing::ImportDistinctFiles, ImportToTrack, frame);
+			}
 		}
 	}
 

libs/ardour/SConscript

@@ -111,7 +111,7 @@ osc_files = [ 'osc.cc' ]
 vst_files = [ 'vst_plugin.cc', 'session_vst.cc' ]
 lv2_files = [ 'lv2_plugin.cc' ]
 audiounit_files = [ 'audio_unit.cc' ]
-coreaudio_files = [ 'coreaudiosource.cc' ]
+coreaudio_files = [ 'coreaudiosource.cc', 'caimportable.cc' ]
 extra_sources = [ ]
 timefx_sources = [ ]
 

libs/ardour/ardour/audiofilesource.h

@@ -168,6 +168,8 @@ class AudioFileSource : public AudioSource {
 	bool removable() const;
 	bool writable() const { return _flags & Writable; }
 
+	static Sample* get_interleave_buffer (nframes_t size);
+
   private:
 	Glib::ustring old_peak_path (Glib::ustring audio_path);
 	Glib::ustring broken_peak_path (Glib::ustring audio_path);

libs/ardour/ardour/caimportable.h

@@ -0,0 +1,48 @@
+/*
+    Copyright (C) 2007 Paul Davis 
+
+    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_ca_importable_source_h__
+#define __ardour_ca_importable_source_h__
+
+#include <pbd/failed_constructor.h>
+#include <ardour/types.h>
+#include <ardour/importable_source.h>
+
+#include <appleutility/CAAudioFile.h>
+
+namespace ARDOUR {
+
+class CAImportableSource : public ImportableSource {
+    public:
+	CAImportableSource (const std::string& path);
+	virtual ~CAImportableSource();
+
+	nframes_t read (Sample* buffer, nframes_t nframes);
+	uint32_t  channels() const;
+	nframes_t length() const;
+	nframes_t samplerate() const;
+	void      seek (nframes_t pos);
+
+   protected:
+	mutable CAAudioFile af;
+};
+
+}
+
+#endif /* __ardour_ca_importable_source_h__ */

libs/ardour/ardour/coreaudiosource.h

@@ -21,7 +21,6 @@
 #define __coreaudio_source_h__
 
 #include <appleutility/CAAudioFile.h>
-
 #include <ardour/audiofilesource.h>
 
 namespace ARDOUR {
@@ -48,11 +47,8 @@ class CoreAudioSource : public AudioFileSource {
 	mutable CAAudioFile af;
 	uint16_t n_channels;
 
-	mutable float *tmpbuf;
-	mutable nframes_t tmpbufsize;
-	mutable Glib::Mutex _tmpbuf_lock;
-
 	void init ();
+	int safe_read (Sample*, nframes_t start, nframes_t cnt, AudioBufferList&) const;
 };
 
 }; /* namespace ARDOUR */

libs/ardour/ardour/sndfilesource.h

@@ -102,8 +102,6 @@ class SndFileSource : public AudioFileSource {
 	void handle_header_position_change ();
 
 	static int64_t get_timecode_info (SNDFILE* sf, SF_BROADCAST_INFO* binfo, bool& exists);
-	
-	static Sample* get_interleave_buffer (nframes_t size);
 };
 
 } // namespace ARDOUR

libs/ardour/audiofilesource.cc

@@ -39,6 +39,7 @@
 
 #include <glibmm/miscutils.h>
 #include <glibmm/fileutils.h>
+#include <glibmm/thread.h>
 
 #include <ardour/audiofilesource.h>
 #include <ardour/sndfile_helpers.h>
@@ -69,6 +70,21 @@ uint64_t           AudioFileSource::header_position_offset = 0;
 /* XXX maybe this too */
 char   AudioFileSource::bwf_serial_number[13] = "000000000000";
 
+struct SizedSampleBuffer {
+    nframes_t size;
+    Sample* buf;
+
+    SizedSampleBuffer (nframes_t sz) : size (sz) { 
+	    buf = new Sample[size];
+    }
+
+    ~SizedSampleBuffer() {
+	    delete [] buf;
+    }
+};
+
+Glib::StaticPrivate<SizedSampleBuffer> thread_interleave_buffer = GLIBMM_STATIC_PRIVATE_INIT;
+
 AudioFileSource::AudioFileSource (Session& s, ustring path, Flag flags)
 	: AudioSource (s, path), _flags (flags),
 	  _channel (0)
@@ -716,3 +732,22 @@ AudioFileSource::mark_immutable ()
 		_flags = Flag (_flags & ~(Writable|Removable|RemovableIfEmpty|RemoveAtDestroy|CanRename));
 	}
 }
+
+
+Sample*
+AudioFileSource::get_interleave_buffer (nframes_t size)
+{
+	SizedSampleBuffer* ssb;
+
+	if ((ssb = thread_interleave_buffer.get()) == 0) {
+		ssb = new SizedSampleBuffer (size);
+		thread_interleave_buffer.set (ssb);
+	}
+
+	if (ssb->size < size) {
+		ssb = new SizedSampleBuffer (size);
+		thread_interleave_buffer.set (ssb);
+	}
+
+	return ssb->buf;
+}

libs/ardour/caimportable.cc

@@ -0,0 +1,118 @@
+#include <ardour/caimportable.h>
+#include <sndfile.h>
+#include <pbd/error.h>
+
+#include "i18n.h"
+
+using namespace ARDOUR;
+using namespace std;
+using namespace PBD;
+
+CAImportableSource::CAImportableSource (const string& path)
+{
+	try {
+		af.Open (path.c_str());
+
+		CAStreamBasicDescription file_format (af.GetFileDataFormat());
+		CAStreamBasicDescription client_format (file_format);
+
+		/* set canonial form (PCM, native float packed, 32 bit, with the correct number of channels
+		   and interleaved (since we plan to deinterleave ourselves)
+		*/
+
+		client_format.SetCanonical(client_format.NumberChannels(), true);
+		af.SetClientFormat (client_format);
+
+	} catch (CAXException& cax) {
+		error << string_compose ("CAImportable: %1", cax.mOperation) << endmsg;
+		throw failed_constructor ();
+	}
+
+}
+
+CAImportableSource::~CAImportableSource ()
+{
+}
+
+nframes_t
+CAImportableSource::read (Sample* buffer, nframes_t nframes) 
+{
+	nframes_t nread = 0;
+	AudioBufferList abl;
+	nframes_t per_channel;
+	bool at_end = false;
+
+	abl.mNumberBuffers = 1;
+	abl.mBuffers[0].mNumberChannels = channels();
+
+	per_channel = nframes / abl.mBuffers[0].mNumberChannels;
+
+	while (nread < per_channel) {
+		
+		UInt32 new_cnt = per_channel - nread;
+		
+		abl.mBuffers[0].mDataByteSize = new_cnt * abl.mBuffers[0].mNumberChannels * sizeof(Sample);
+		abl.mBuffers[0].mData = buffer + nread;
+			
+		try {
+			af.Read (new_cnt, &abl);
+		} catch (CAXException& cax) {
+			error << string_compose("CAImportable: %1", cax.mOperation);
+			return -1;
+		}
+
+		if (new_cnt == 0) {
+			/* EOF */
+			at_end = true;
+			break;
+		}
+
+		nread += new_cnt;
+	}
+
+	if (!at_end && nread < per_channel) {
+		return 0;
+	} else {
+		return nread * 	abl.mBuffers[0].mNumberChannels;
+	}
+}
+
+uint
+CAImportableSource::channels () const 
+{
+	return af.GetFileDataFormat().NumberChannels();
+}
+
+nframes_t
+CAImportableSource::length () const 
+{
+	return af.GetNumberFrames();
+}
+
+nframes_t
+CAImportableSource::samplerate() const
+{
+	CAStreamBasicDescription client_asbd;
+
+	try {
+		client_asbd = af.GetClientDataFormat ();
+	} catch (CAXException& cax) {
+		error << string_compose ("CAImportable: %1", cax.mOperation) << endmsg;
+		return 0.0;
+	}
+
+	return client_asbd.mSampleRate;
+}
+
+void
+CAImportableSource::seek (nframes_t pos)
+{
+	try {
+		af.Seek (pos);
+	} catch (CAXException& cax) {
+		error << string_compose ("CAImportable: %1 to %2", cax.mOperation, pos) << endmsg;
+	}
+}
+
+	
+

libs/ardour/coreaudiosource.cc

@@ -1,5 +1,6 @@
 /*
-    Copyright (C) 2006 Paul Davis 
+    Copyright (C) 2006 Paul Davis
+    Written by Taybin Rutkin
 
     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
@@ -17,6 +18,8 @@
 
 */
 
+#include <algorithm>
+
 #include <pbd/error.h>
 #include <ardour/coreaudiosource.h>
 #include <ardour/utils.h>
@@ -28,6 +31,7 @@
 
 #include <AudioToolbox/AudioFormat.h>
 
+using namespace std;
 using namespace ARDOUR;
 using namespace PBD;
 
@@ -48,18 +52,12 @@ CoreAudioSource::CoreAudioSource (Session& s, const string& path, int chn, Flag
 void 
 CoreAudioSource::init ()
 {
-	tmpbuf = 0;
-	tmpbufsize = 0;
-
-	cerr << "CoreAudioSource::init() " << name() << endl;
-	
 	/* note that we temporarily truncated _id at the colon */
 	try {
 		af.Open(_path.c_str());
 
-		CAStreamBasicDescription file_asbd (af.GetFileDataFormat());
+		CAStreamBasicDescription file_format (af.GetFileDataFormat());
-		n_channels = file_asbd.NumberChannels();
+		n_channels = file_format.NumberChannels();
-		cerr << "number of channels: " << n_channels << endl;
 		
 		if (_channel >= n_channels) {
 			error << string_compose("CoreAudioSource: file only contains %1 channels; %2 is invalid as a channel number (%3)", n_channels, _channel, name()) << endmsg;
@@ -68,9 +66,15 @@ CoreAudioSource::init ()
 
 		_length = af.GetNumberFrames();
 
-		CAStreamBasicDescription client_asbd(file_asbd);
+		CAStreamBasicDescription client_format (file_format);
-		client_asbd.SetCanonical(client_asbd.NumberChannels(), false);
+
-		af.SetClientFormat (client_asbd);
+		/* set canonial form (PCM, native float packed, 32 bit, with the correct number of channels
+		   and interleaved (since we plan to deinterleave ourselves)
+		*/
+
+		client_format.SetCanonical(client_format.NumberChannels(), true);
+		af.SetClientFormat (client_format);
+
 	} catch (CAXException& cax) {
 		error << string_compose ("CoreAudioSource: %1 (%2)", cax.mOperation, name()) << endmsg;
 		throw failed_constructor ();
@@ -79,81 +83,117 @@ CoreAudioSource::init ()
 
 CoreAudioSource::~CoreAudioSource ()
 {
-	cerr << "CoreAudioSource::~CoreAudioSource() " << name() << endl;
 	GoingAway (); /* EMIT SIGNAL */
-
-	if (tmpbuf) {
-		delete [] tmpbuf;
-	}
-	
-	cerr << "deletion done" << endl;
 }
 
-nframes_t
+int
-CoreAudioSource::read_unlocked (Sample *dst, nframes_t start, nframes_t cnt) const
+CoreAudioSource::safe_read (Sample* dst, nframes_t start, nframes_t cnt, AudioBufferList& abl) const
 {
-	try {
+	nframes_t nread = 0;
-		af.Seek (start);
-	} catch (CAXException& cax) {
-		error << string_compose("CoreAudioSource: %1 to %2 (%3)", cax.mOperation, start, _name.substr (1)) << endmsg;
-		return 0;
-	}
 
-	AudioBufferList abl;
+	while (nread < cnt) {
-	abl.mNumberBuffers = 1;
+		
-	abl.mBuffers[0].mNumberChannels = n_channels;
+		try {
-
+			af.Seek (start+nread);
-	UInt32 new_cnt = cnt;
+		} catch (CAXException& cax) {
-	if (n_channels == 1) {
+			error << string_compose("CoreAudioSource: %1 to %2 (%3)", cax.mOperation, start+nread, _name.substr (1)) << endmsg;
-		abl.mBuffers[0].mDataByteSize = cnt * sizeof(Sample);
+			return -1;
-		abl.mBuffers[0].mData = dst;
+		}
+		
+		UInt32 new_cnt = cnt - nread;
+		
+		abl.mBuffers[0].mDataByteSize = new_cnt * n_channels * sizeof(Sample);
+		abl.mBuffers[0].mData = dst + nread;
+			
 		try {
 			af.Read (new_cnt, &abl);
 		} catch (CAXException& cax) {
 			error << string_compose("CoreAudioSource: %1 (%2)", cax.mOperation, _name);
+			return -1;
 		}
-		_read_data_count = new_cnt * sizeof(float);
+
-		return new_cnt;
+		if (new_cnt == 0) {
+			/* EOF */
+			if (start+cnt == _length) {
+				/* we really did hit the end */
+				nread = cnt;
+			}
+			break;
+		}
+
+		nread += new_cnt;
 	}
 
-	UInt32 real_cnt = cnt * n_channels;
+	if (nread < cnt) {
+		return -1;
+	} else {
+		return 0;
+	}
+}
+	
 
-	{
+nframes_t
-		Glib::Mutex::Lock lm (_tmpbuf_lock);
+CoreAudioSource::read_unlocked (Sample *dst, nframes_t start, nframes_t cnt) const
+{
+	nframes_t file_cnt;
+	AudioBufferList abl;
+
+	abl.mNumberBuffers = 1;
+	abl.mBuffers[0].mNumberChannels = n_channels;
+
+	if (start > _length) {
+
+		/* read starts beyond end of data, just memset to zero */
 		
-		if (tmpbufsize < real_cnt) {
+		file_cnt = 0;
-			
+
-			if (tmpbuf) {
+	} else if (start + cnt > _length) {
-				delete [] tmpbuf;
+		
+		/* read ends beyond end of data, read some, memset the rest */
+		
+		file_cnt = _length - start;
+
+	} else {
+		
+		/* read is entirely within data */
+
+		file_cnt = cnt;
+	}
+
+	if (file_cnt != cnt) {
+		nframes_t delta = cnt - file_cnt;
+		memset (dst+file_cnt, 0, sizeof (Sample) * delta);
+	}
+
+	if (file_cnt) {
+
+		if (n_channels == 1) {
+			if (safe_read (dst, start, file_cnt, abl) == 0) {
+				_read_data_count = cnt * sizeof (Sample);
+				return cnt;
 			}
-			tmpbufsize = real_cnt;
+			return 0;
-			tmpbuf = new float[tmpbufsize];
 		}
+	}
 
-		abl.mBuffers[0].mDataByteSize = tmpbufsize * sizeof(Sample);
+	Sample* interleave_buf = get_interleave_buffer (file_cnt * n_channels);
-		abl.mBuffers[0].mData = tmpbuf;
+	
-
+	if (safe_read (interleave_buf, start, file_cnt, abl) != 0) {
-		cerr << "channel: " << _channel << endl;
+		return 0;
-		
-		try {
-			af.Read (real_cnt, &abl);
-		} catch (CAXException& cax) {
-			error << string_compose("CoreAudioSource: %1 (%2)", cax.mOperation, _name);
-		}
-		float *ptr = tmpbuf + _channel;
-		real_cnt /= n_channels;
-		
-		/* stride through the interleaved data */
-		
-		for (uint32_t n = 0; n < real_cnt; ++n) {
-			dst[n] = *ptr;
-			ptr += n_channels;
-		}
 	}
 
 	_read_data_count = cnt * sizeof(float);
-		
+
-	return real_cnt;
+	Sample *ptr = interleave_buf + _channel;
+	
+	/* stride through the interleaved data */
+	
+	for (uint32_t n = 0; n < file_cnt; ++n) {
+		dst[n] = *ptr;
+		ptr += n_channels;
+	}
+
+	return cnt;
 }
 
 float

libs/ardour/playlist.cc

@@ -334,7 +334,6 @@ Playlist::release_notifications ()
 	} 
 }
 
-
 void
 Playlist::notify_modified ()
 {

libs/ardour/resampled_source.cc

@@ -26,7 +26,7 @@
 using namespace ARDOUR;
 using namespace PBD;
 
-const uint32_t ResampledImportableSource::blocksize = 4096U;
+const uint32_t ResampledImportableSource::blocksize = 16384U;
 
 ResampledImportableSource::ResampledImportableSource (boost::shared_ptr<ImportableSource> src, nframes_t rate, SrcQuality srcq)
 	: source (src)

libs/ardour/session.cc

@@ -482,7 +482,7 @@ Session::destroy ()
 
 	// auditioner.reset ();
 	
-#undef TRACK_DESTRUCTION
+#define TRACK_DESTRUCTION
 #ifdef TRACK_DESTRUCTION
 	cerr << "delete named selections\n";
 #endif /* TRACK_DESTRUCTION */
@@ -577,14 +577,19 @@ Session::destroy ()
 
 		tmp = i;
 		++tmp;
+		
+		cerr << "Drop refs to " << i->second->name() << endl;
 
 		i->second->drop_references ();
 
+		cerr << "move on\n";
+		
 		i = tmp;
 	}
-
+	
+	cerr << "clear audio sources\n";
 	audio_sources.clear ();
-
+	
 #ifdef TRACK_DESTRUCTION
 	cerr << "delete mix groups\n";
 #endif /* TRACK_DESTRUCTION */
@@ -2892,6 +2897,8 @@ Session::remove_source (boost::weak_ptr<Source> src)
 		return;
 	} 
 
+	cerr << "remove source for " << source->name() << endl;
+
 	{ 
 		Glib::Mutex::Lock lm (audio_source_lock);
 		

libs/ardour/sndfilesource.cc

@@ -25,7 +25,6 @@
 #include <sys/stat.h>
 
 #include <glibmm/miscutils.h>
-#include <glibmm/thread.h>
 #include <pbd/stacktrace.h>
 
 #include <ardour/sndfilesource.h>
@@ -46,21 +45,6 @@ const AudioFileSource::Flag SndFileSource::default_writable_flags = AudioFileSou
 											   AudioFileSource::Removable|
 											   AudioFileSource::RemovableIfEmpty|
 											   AudioFileSource::CanRename);
-struct SizedSampleBuffer {
-    nframes_t size;
-    Sample* buf;
-
-    SizedSampleBuffer (nframes_t sz) : size (sz) { 
-	    buf = new Sample[size];
-    }
-
-    ~SizedSampleBuffer() {
-	    delete [] buf;
-    }
-};
-
-Glib::StaticPrivate<SizedSampleBuffer> thread_interleave_buffer = GLIBMM_STATIC_PRIVATE_INIT;
-
 SndFileSource::SndFileSource (Session& s, const XMLNode& node)
 	: AudioFileSource (s, node)
 {
@@ -96,7 +80,7 @@ SndFileSource::SndFileSource (Session& s, ustring path, SampleFormat sfmt, Heade
 	*/
 
 	file_is_new = true;
-	
+
 	switch (hf) {
 	case CAF:
 		fmt = SF_FORMAT_CAF;
@@ -190,9 +174,8 @@ SndFileSource::SndFileSource (Session& s, ustring path, SampleFormat sfmt, Heade
 			_flags = Flag (_flags & ~Broadcast);
 			delete _broadcast_info;
 			_broadcast_info = 0;
-		}
+		} 
-		
+	} 
-	}
 }
 
 void 
@@ -256,7 +239,7 @@ SndFileSource::open ()
 
 	set_timeline_position (get_timecode_info (sf, _broadcast_info, timecode_info_exists));
 
-	if (!timecode_info_exists) {
+	if (_length != 0 && !timecode_info_exists) {
 		delete _broadcast_info;
 		_broadcast_info = 0;
 		_flags = Flag (_flags & ~Broadcast);
@@ -328,6 +311,11 @@ SndFileSource::read_unlocked (Sample *dst, nframes_t start, nframes_t cnt) const
 		file_cnt = cnt;
 	}
 	
+	if (file_cnt != cnt) {
+		nframes_t delta = cnt - file_cnt;
+		memset (dst+file_cnt, 0, sizeof (Sample) * delta);
+	}
+
 	if (file_cnt) {
 
 		if (sf_seek (sf, (sf_count_t) start, SEEK_SET|SFM_READ) != (sf_count_t) start) {
@@ -344,11 +332,6 @@ SndFileSource::read_unlocked (Sample *dst, nframes_t start, nframes_t cnt) const
 		}
 	}
 
-	if (file_cnt != cnt) {
-		nframes_t delta = cnt - file_cnt;
-		memset (dst+file_cnt, 0, sizeof (Sample) * delta);
-	}
-
 	real_cnt = cnt * _info.channels;
 
 	Sample* interleave_buf = get_interleave_buffer (real_cnt);
@@ -908,20 +891,3 @@ SndFileSource::one_of_several_channels () const
 	return _info.channels > 1;
 }
 
-Sample*
-SndFileSource::get_interleave_buffer (nframes_t size)
-{
-	SizedSampleBuffer* ssb;
-
-	if ((ssb = thread_interleave_buffer.get()) == 0) {
-		ssb = new SizedSampleBuffer (size);
-		thread_interleave_buffer.set (ssb);
-	}
-
-	if (ssb->size < size) {
-		ssb = new SizedSampleBuffer (size);
-		thread_interleave_buffer.set (ssb);
-	}
-
-	return ssb->buf;
-}

libs/ardour/source_factory.cc

@@ -30,12 +30,15 @@
 #include <ardour/silentfilesource.h>
 #include <ardour/configuration.h>
 
-#undef USE_COREAUDIO_FOR_FILES
+#ifdef  HAVE_COREAUDIO
+#define USE_COREAUDIO_FOR_FILES
+#endif
 
 #ifdef USE_COREAUDIO_FOR_FILES
 #include <ardour/coreaudiosource.h>
 #endif
 
+
 #include "i18n.h"
 
 using namespace ARDOUR;
@@ -73,6 +76,7 @@ peak_thread_work ()
 		if (!as) {
 			continue;
 		}
+
 		as->setup_peakfile ();
 	}
 }
@@ -121,12 +125,12 @@ SourceFactory::createSilent (Session& s, const XMLNode& node, nframes_t nframes,
 	return ret;
 }
 
-#ifdef USE_COREAUDIO_FOR_FILES
 boost::shared_ptr<Source>
 SourceFactory::create (Session& s, const XMLNode& node, bool defer_peaks)
 {
 	try {
-		boost::shared_ptr<Source> ret (new CoreAudioSource (s, node));
+
+		boost::shared_ptr<Source> ret (new SndFileSource (s, node));
 		if (setup_peakfile (ret, defer_peaks)) {
 			return boost::shared_ptr<Source>();
 		}
@@ -138,47 +142,34 @@ SourceFactory::create (Session& s, const XMLNode& node, bool defer_peaks)
 
 	catch (failed_constructor& err) {	
 
+#ifdef USE_COREAUDIO_FOR_FILES
+
 		/* this is allowed to throw */
 
-		boost::shared_ptr<Source> ret (new SndFileSource (s, node));
+		boost::shared_ptr<Source> ret (new CoreAudioSource (s, node));
 		if (setup_peakfile (ret, defer_peaks)) {
 			return boost::shared_ptr<Source>();
 		}
 		ret->check_for_analysis_data_on_disk ();
 		SourceCreated (ret);
 		return ret;
-	}
-
-	return boost::shared_ptr<Source>();
-}
-
 #else
+		throw; // rethrow 
+#endif
 
-boost::shared_ptr<Source>
-SourceFactory::create (Session& s, const XMLNode& node, bool defer_peaks)
-{
-	/* this is allowed to throw */
-
-	boost::shared_ptr<Source> ret (new SndFileSource (s, node));
-	
-	if (setup_peakfile (ret, defer_peaks)) {
-		return boost::shared_ptr<Source>();
 	}
-	ret->check_for_analysis_data_on_disk ();
+
-	SourceCreated (ret);
+	return boost::shared_ptr<Source>(); // keep stupid gcc happy
-	return ret;
 }
 
-#endif // USE_COREAUDIO_FOR_FILES
-
-#ifdef USE_COREAUDIO_FOR_FILES
 boost::shared_ptr<Source>
 SourceFactory::createReadable (Session& s, string path, int chn, AudioFileSource::Flag flags, bool announce, bool defer_peaks)
 {
 	if (!(flags & Destructive)) {
 
 		try {
-			boost::shared_ptr<Source> ret (new CoreAudioSource (s, path, chn, flags));
+
+			boost::shared_ptr<Source> ret (new SndFileSource (s, path, chn, flags));
 			if (setup_peakfile (ret, defer_peaks)) {
 				return boost::shared_ptr<Source>();
 			}
@@ -193,7 +184,9 @@ SourceFactory::createReadable (Session& s, string path, int chn, AudioFileSource
 
 			/* this is allowed to throw */
 
-			boost::shared_ptr<Source> ret (new SndFileSource (s, path, chn, flags));
+#ifdef USE_COREAUDIO_FOR_FILES
+
+			boost::shared_ptr<Source> ret (new CoreAudioSource (s, path, chn, flags));
 			if (setup_peakfile (ret, defer_peaks)) {
 				return boost::shared_ptr<Source>();
 			}
@@ -202,46 +195,19 @@ SourceFactory::createReadable (Session& s, string path, int chn, AudioFileSource
 				SourceCreated (ret);
 			}
 			return ret;
+
+#else
+			throw; // rethrow
+#endif
 		}
 
 	} else {
 
-		boost::shared_ptr<Source> ret (new SndFileSource (s, path, chn, flags));
-		if (setup_peakfile (ret, defer_peaks)) {
-			return boost::shared_ptr<Source>();
-		}
-		ret->check_for_analysis_data_on_disk ();
-		if (announce) {
-			SourceCreated (ret);
-		}
-		return ret;
 	}
 
 	return boost::shared_ptr<Source>();
 }
 
-#else
-
-boost::shared_ptr<Source>
-SourceFactory::createReadable (Session& s, string path, int chn, AudioFileSource::Flag flags, bool announce, bool defer_peaks)
-{
-	boost::shared_ptr<Source> ret (new SndFileSource (s, path, chn, flags));
-
-	if (setup_peakfile (ret, defer_peaks)) {
-		return boost::shared_ptr<Source>();
-	}
-
-	ret->check_for_analysis_data_on_disk ();
-
-	if (announce) {
-		SourceCreated (ret);
-	}
-
-	return ret;
-}
-
-#endif // USE_COREAUDIO_FOR_FILES
-
 boost::shared_ptr<Source>
 SourceFactory::createWritable (Session& s, std::string path, bool destructive, nframes_t rate, bool announce, bool defer_peaks)
 {

svn_revision.h

@@ -1,4 +1,4 @@
 #ifndef __ardour_svn_revision_h__
 #define __ardour_svn_revision_h__
-static const char* ardour_svn_revision = "3107";
+static const char* ardour_svn_revision = "3117";
 #endif