merge with 2.0-ongoing @ rev 3147

git-svn-id: svn://localhost/ardour2/branches/3.0@3152 d708f5d6-7413-0410-9779-e7cbd77b26cf

1 files changed, 108 insertions, 66 deletions

diff --git a/libs/ardour/coreaudiosource.cc b/libs/ardour/coreaudiosource.cc
index 703225e502..4383f1a696 100644
--- a/libs/ardour/coreaudiosource.cc
+++ b/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());
-		n_channels = file_asbd.NumberChannels();
-		cerr << "number of channels: " << n_channels << endl;
+		CAStreamBasicDescription file_format (af.GetFileDataFormat());
+		n_channels = file_format.NumberChannels();
 		
 		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;
@@ -68,92 +66,136 @@ CoreAudioSource::init ()
 
 		_length = af.GetNumberFrames();
 
-		CAStreamBasicDescription client_asbd(file_asbd);
-		client_asbd.SetCanonical(client_asbd.NumberChannels(), false);
-		af.SetClientFormat (client_asbd);
+		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 ("CoreAudioSource: %1 (%2)", cax.mOperation, name()) << endmsg;
+		
+		error << string_compose(_("CoreAudioSource: cannot open file \"%1\" for %2"), 
+					_path, (writable() ? "read+write" : "reading")) << endmsg;
 		throw failed_constructor ();
 	}
 }
 
 CoreAudioSource::~CoreAudioSource ()
 {
-	cerr << "CoreAudioSource::~CoreAudioSource() " << name() << endl;
 	GoingAway (); /* EMIT SIGNAL */
-
-	if (tmpbuf) {
-		delete [] tmpbuf;
-	}
-	
-	cerr << "deletion done" << endl;
 }
 
-nframes_t
-CoreAudioSource::read_unlocked (Sample *dst, nframes_t start, nframes_t cnt) const
+int
+CoreAudioSource::safe_read (Sample* dst, nframes_t start, nframes_t cnt, AudioBufferList& abl) const
 {
-	try {
-		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;
-	abl.mNumberBuffers = 1;
-	abl.mBuffers[0].mNumberChannels = n_channels;
+	nframes_t nread = 0;
 
-	UInt32 new_cnt = cnt;
-	if (n_channels == 1) {
-		abl.mBuffers[0].mDataByteSize = cnt * sizeof(Sample);
-		abl.mBuffers[0].mData = dst;
+	while (nread < cnt) {
+		
+		try {
+			af.Seek (start+nread);
+		} catch (CAXException& cax) {
+			error << string_compose("CoreAudioSource: %1 to %2 (%3)", cax.mOperation, start+nread, _name.substr (1)) << endmsg;
+			return -1;
+		}
+		
+		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;
-	}
-
-	UInt32 real_cnt = cnt * n_channels;
 
-	{
-		Glib::Mutex::Lock lm (_tmpbuf_lock);
-		
-		if (tmpbufsize < real_cnt) {
-			
-			if (tmpbuf) {
-				delete [] tmpbuf;
+		if (new_cnt == 0) {
+			/* EOF */
+			if (start+cnt == _length) {
+				/* we really did hit the end */
+				nread = cnt;
 			}
-			tmpbufsize = real_cnt;
-			tmpbuf = new float[tmpbufsize];
+			break;
 		}
 
-		abl.mBuffers[0].mDataByteSize = tmpbufsize * sizeof(Sample);
-		abl.mBuffers[0].mData = tmpbuf;
+		nread += new_cnt;
+	}
 
-		cerr << "channel: " << _channel << endl;
+	if (nread < cnt) {
+		return -1;
+	} else {
+		return 0;
+	}
+}
+	
+
+nframes_t
+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 */
 		
-		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;
+		file_cnt = 0;
+
+	} else if (start + cnt > _length) {
 		
-		/* stride through the interleaved data */
+		/* read ends beyond end of data, read some, memset the rest */
+		
+		file_cnt = _length - start;
+
+	} else {
 		
-		for (uint32_t n = 0; n < real_cnt; ++n) {
-			dst[n] = *ptr;
-			ptr += n_channels;
+		/* 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;
+			}
+			return 0;
 		}
 	}
 
+	Sample* interleave_buf = get_interleave_buffer (file_cnt * n_channels);
+	
+	if (safe_read (interleave_buf, start, file_cnt, abl) != 0) {
+		return 0;
+	}
+
 	_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