Version 4.3.1

src/FileRead.cpp

@@ -39,10 +39,11 @@ FileRead :: FileRead()
 {
 }
 
-FileRead :: FileRead( std::string fileName, bool typeRaw )
+FileRead :: FileRead( std::string fileName, bool typeRaw, unsigned int nChannels,
+                      StkFormat format, StkFloat rate )
   : fd_(0)
 {
-  open( fileName, typeRaw );
+  open( fileName, typeRaw, nChannels, format, rate );
 }
 
 FileRead :: ~FileRead()
@@ -64,7 +65,8 @@ bool FileRead :: isOpen( void )
   else return false;
 }
 
-void FileRead :: open( std::string fileName, bool typeRaw )
+void FileRead :: open( std::string fileName, bool typeRaw, unsigned int nChannels,
+                       StkFormat format, StkFloat rate )
 {
   // If another file is open, close it.
   close();
@@ -79,7 +81,7 @@ void FileRead :: open( std::string fileName, bool typeRaw )
   // Attempt to determine file type from header (unless RAW).
   bool result = false;
   if ( typeRaw )
-    result = getRawInfo( fileName.c_str() );
+    result = getRawInfo( fileName.c_str(), nChannels, format, rate );
   else {
     char header[12];
     if ( fread( &header, 4, 3, fd_ ) != 3 ) goto error;
@@ -121,7 +123,7 @@ void FileRead :: open( std::string fileName, bool typeRaw )
   handleError( StkError::FILE_ERROR );
 }
 
-bool FileRead :: getRawInfo( const char *fileName )
+bool FileRead :: getRawInfo( const char *fileName, unsigned int nChannels, StkFormat format, StkFloat rate )
 {
   // Use the system call "stat" to determine the file length.
   struct stat filestat;
@@ -130,14 +132,22 @@ bool FileRead :: getRawInfo( const char *fileName )
     return false;
   }
 
-  // STK rawwave files have no header and are assumed to contain a
-  // monophonic stream of 16-bit signed integers in big-endian byte
-  // order at a sample rate of 22050 Hz.
-  channels_ = 1;
-  fileSize_ = (long) filestat.st_size / 2;  // length in 2-byte samples
+  // Rawwave files have no header and by default, are assumed to
+  // contain a monophonic stream of 16-bit signed integers in
+  // big-endian byte order at a sample rate of 22050 Hz.  However,
+  // different parameters can be specified if desired.
   dataOffset_ = 0;
-  fileRate_ = 22050.0;
-  dataType_ = STK_SINT16;
+  channels_ = nChannels;
+  dataType_ = format;
+  fileRate_ = rate;
+  int sampleBytes = 0;
+  if ( format == STK_SINT8 ) sampleBytes = 1;
+  else if ( format == STK_SINT16 ) sampleBytes = 2;
+  else if ( format == STK_SINT32 || format == STK_FLOAT32 ) sampleBytes = 4;
+  else if ( format == STK_FLOAT64 ) sampleBytes = 8;
+
+  fileSize_ = (long) filestat.st_size / sampleBytes / channels_;  // length in frames
+
   byteswap_ = false;
 #ifdef __LITTLE_ENDIAN__
   byteswap_ = true;

src/Makefile.in

@@ -1,3 +1,4 @@
+### Do not edit -- Generated by 'configure --with-whatever' from Makefile.in
 ### libstk Makefile - for various flavors of unix
 
 LIBRARY = libstk.a
@@ -38,11 +39,11 @@ CC       = @CXX@
 DEFS     = @byte_order@
 DEFS    += @debug@
 CFLAGS   = @cflags@
-CFLAGS  += @warn@ $(INCLUDE)
+CFLAGS  += @warn@ $(INCLUDE) -Iinclude
 
 REALTIME = @realtime@
 ifeq ($(REALTIME),yes)
-	OBJECTS += RtMidi.o RtAudio.o RtWvOut.o RtWvIn.o InetWvOut.o InetWvIn.o Thread.o Mutex.o Socket.o TcpClient.o TcpServer.o UdpSocket.o
+	OBJECTS += RtMidi.o RtAudio.o RtWvOut.o RtWvIn.o InetWvOut.o InetWvIn.o Thread.o Mutex.o Socket.o TcpClient.o TcpServer.o UdpSocket.o @objects@
 	DEFS    += @audio_apis@
 endif
 
@@ -55,6 +56,9 @@ DEFS    += -DRAWWAVE_PATH=\"$(RAWWAVES)\"
 %.o : %.cpp
 	$(CC) $(CFLAGS) $(DEFS) -c $(<) -o $(OBJECT_PATH)/$@
 
+%.o : ../src/include/%.cpp
+	$(CC) $(CFLAGS) $(DEFS) -c $(<) -o $(OBJECT_PATH)/$@
+
 all : $(LIBRARY)
 
 $(LIBRARY) : $(OBJECTS)

src/RtAudio.cpp

@@ -38,7 +38,7 @@
 */
 /************************************************************************/
 
-// RtAudio: Version 4.0
+// RtAudio: Version 4.0.3
 
 #include "RtAudio.h"
 #include <iostream>
@@ -539,7 +539,7 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
     error( RtError::WARNING );
     return info;
   }
-  info.name.append( (const char *)name, strlen(name) + 1 );
+  info.name.append( (const char *)name, strlen(name) );
   info.name.append( ": " );
 
   dataSize = 256;
@@ -552,7 +552,7 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
     error( RtError::WARNING );
     return info;
   }
-  info.name.append( (const char *)name, strlen(name) + 1 );
+  info.name.append( (const char *)name, strlen(name) );
 
   // Get the output stream "configuration".
   AudioBufferList	*bufferList = nil;
@@ -2291,10 +2291,10 @@ bool RtApiJack :: callbackEvent( unsigned long nframes )
 // on information found in
 // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
 
-#include "asio/asiosys.h"
-#include "asio/asio.h"
-#include "asio/iasiothiscallresolver.h"
-#include "asio/asiodrivers.h"
+#include "asiosys.h"
+#include "asio.h"
+#include "iasiothiscallresolver.h"
+#include "asiodrivers.h"
 #include <cmath>
 
 AsioDrivers drivers;
@@ -2366,11 +2366,14 @@ RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
     error( RtError::INVALID_USE );
   }
 
-  // Don't probe if a stream is already open.
+  // If a stream is already open, we cannot probe other devices.  Thus, use the saved results.
   if ( stream_.state != STREAM_CLOSED ) {
-    errorText_ = "RtApiAsio::getDeviceInfo: unable to probe driver while a stream is open.";
-    error( RtError::WARNING );
-    return info;
+    if ( device >= devices_.size() ) {
+      errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
+      error( RtError::WARNING );
+      return info;
+    }
+    return devices_[ device ];
   }
 
   char driverName[32];
@@ -2463,6 +2466,16 @@ void bufferSwitch( long index, ASIOBool processNow )
   object->callbackEvent( index );
 }
 
+void RtApiAsio :: saveDeviceInfo( void )
+{
+  devices_.clear();
+
+  unsigned int nDevices = getDeviceCount();
+  devices_.resize( nDevices );
+  for ( unsigned int i=0; i<nDevices; i++ )
+    devices_[i] = getDeviceInfo( i );
+}
+
 bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
                                    unsigned int firstChannel, unsigned int sampleRate,
                                    RtAudioFormat format, unsigned int *bufferSize,
@@ -2482,6 +2495,12 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
     return FAILURE;
   }
 
+  // The getDeviceInfo() function will not work when a stream is open
+  // because ASIO does not allow multiple devices to run at the same
+  // time.  Thus, we'll probe the system before opening a stream and
+  // save the results for use by getDeviceInfo().
+  this->saveDeviceInfo();
+
   // Only load the driver once for duplex stream.
   if ( mode != INPUT || stream_.mode != OUTPUT ) {
     if ( !drivers.loadDriver( driverName ) ) {
@@ -2528,15 +2547,27 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
     return FAILURE;
   }
 
-  // Set the sample rate.
-  result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
+  // Get the current sample rate
+  ASIOSampleRate currentRate;
+  result = ASIOGetSampleRate( &currentRate );
   if ( result != ASE_OK ) {
     drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
     errorText_ = errorStream_.str();
     return FAILURE;
   }
 
+  // Set the sample rate only if necessary
+  if ( currentRate != sampleRate ) {
+    result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
+    if ( result != ASE_OK ) {
+      drivers.removeCurrentDriver();
+      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+  }
+
   // Determine the driver data type.
   ASIOChannelInfo channelInfo;
   channelInfo.channel = 0;
@@ -2696,7 +2727,8 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
     stream_.doConvertBuffer[mode] = true;
 
   // Allocate necessary internal buffers
-  unsigned long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+  unsigned long bufferBytes;
+  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
   stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
   if ( stream_.userBuffer[mode] == NULL ) {
     errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
@@ -2958,8 +2990,8 @@ bool RtApiAsio :: callbackEvent( long bufferIndex )
       handle->internalDrain = true;
   }
 
-  unsigned int bufferBytes, i, j;
-  unsigned int nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+  unsigned int nChannels, bufferBytes, i, j;
+  nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
 
     bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
@@ -3187,6 +3219,14 @@ static const char* getAsioErrorString( ASIOError result )
 #include <dsound.h>
 #include <assert.h>
 
+#if defined(__MINGW32__)
+// missing from latest mingw winapi
+#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
+#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
+#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
+#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
+#endif
+
 #define MINIMUM_DEVICE_BUFFER_SIZE 32768
 
 #ifdef _MSC_VER // if Microsoft Visual C++
@@ -3223,7 +3263,7 @@ struct DsHandle {
   HANDLE condition;
 
   DsHandle()
-    :drainCounter(0), internalDrain(false) { id[0] = 0, id[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
+    :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
 };
 
 /*
@@ -3247,7 +3287,7 @@ RtApiDs::RtDsStatistics RtApiDs::getDsStatistics()
 
 // Declarations for utility functions, callbacks, and structures
 // specific to the DirectSound implementation.
-static bool CALLBACK deviceCountCallback( LPGUID lpguid,
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
                                           LPCTSTR description,
                                           LPCTSTR module,
                                           LPVOID lpContext );
@@ -3288,7 +3328,7 @@ unsigned int RtApiDs :: getDefaultInputDevice( void )
 {
   // Count output devices.
   EnumInfo info;
-  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &info );
+  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
   if ( FAILED( result ) ) {
     errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!";
     errorText_ = errorStream_.str();
@@ -3299,7 +3339,7 @@ unsigned int RtApiDs :: getDefaultInputDevice( void )
   // Now enumerate input devices until we find the id = NULL.
   info.isInput = true;
   info.getDefault = true;
-  result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &info );
+  result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
   if ( FAILED( result ) ) {
     errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!";
     errorText_ = errorStream_.str();
@@ -3316,7 +3356,7 @@ unsigned int RtApiDs :: getDefaultOutputDevice( void )
   // Enumerate output devices until we find the id = NULL.
   EnumInfo info;
   info.getDefault = true;
-  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &info );
+  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
   if ( FAILED( result ) ) {
     errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!";
     errorText_ = errorStream_.str();
@@ -3332,7 +3372,7 @@ unsigned int RtApiDs :: getDeviceCount( void )
 {
   // Count DirectSound devices.
   EnumInfo info;
-  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &info );
+  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
   if ( FAILED( result ) ) {
     errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
     errorText_ = errorStream_.str();
@@ -3341,7 +3381,7 @@ unsigned int RtApiDs :: getDeviceCount( void )
 
   // Count DirectSoundCapture devices.
   info.isInput = true;
-  result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &info );
+  result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
   if ( FAILED( result ) ) {
     errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
     errorText_ = errorStream_.str();
@@ -3366,7 +3406,7 @@ RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
   EnumInfo dsinfo;
   dsinfo.findIndex = true;
   dsinfo.index = device;
-  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &dsinfo );
+  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
   if ( FAILED( result ) ) {
     errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating output devices!";
     errorText_ = errorStream_.str();
@@ -3424,7 +3464,7 @@ RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
  probeInput:
 
   dsinfo.isInput = true;
-  result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &dsinfo );
+  result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
   if ( FAILED( result ) ) {
     errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!";
     errorText_ = errorStream_.str();
@@ -3534,7 +3574,7 @@ bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned
   EnumInfo dsinfo;
   dsinfo.findIndex = true;
   dsinfo.index = device;
-  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &dsinfo );
+  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
   if ( FAILED( result ) ) {
     errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!";
     errorText_ = errorStream_.str();
@@ -3550,7 +3590,7 @@ bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned
   }
   else { // mode == INPUT
     dsinfo.isInput = true;
-    HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceCountCallback, &dsinfo );
+    HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
     if ( FAILED( result ) ) {
       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!";
       errorText_ = errorStream_.str();
@@ -3858,6 +3898,7 @@ bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned
   }
 
   // Set various stream parameters
+  DsHandle *handle = 0;
   stream_.nDeviceChannels[mode] = channels + firstChannel;
   stream_.nUserChannels[mode] = channels;
   stream_.bufferSize = *bufferSize;
@@ -3907,7 +3948,6 @@ bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned
   }
 
   // Allocate our DsHandle structures for the stream.
-  DsHandle *handle;
   if ( stream_.apiHandle == 0 ) {
     try {
       handle = new DsHandle;
@@ -4078,7 +4118,7 @@ void RtApiDs :: startStream()
     duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
   }
 
-  HRESULT result;
+  HRESULT result = 0;
   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
     //statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[0];
 
@@ -4124,7 +4164,7 @@ void RtApiDs :: stopStream()
 
   MUTEX_LOCK( &stream_.mutex );
 
-  HRESULT result;
+  HRESULT result = 0;
   LPVOID audioPtr;
   DWORD dataLen;
   DsHandle *handle = (DsHandle *) stream_.apiHandle;
@@ -4665,7 +4705,7 @@ std::string convertTChar( LPCTSTR name )
   return s;
 }
 
-static bool CALLBACK deviceCountCallback( LPGUID lpguid,
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
                                           LPCTSTR description,
                                           LPCTSTR module,
                                           LPVOID lpContext )
@@ -4678,7 +4718,7 @@ static bool CALLBACK deviceCountCallback( LPGUID lpguid,
     LPDIRECTSOUNDCAPTURE object;
 
     hr = DirectSoundCaptureCreate(  lpguid, &object,   NULL );
-    if ( hr != DS_OK ) return true;
+    if ( hr != DS_OK ) return TRUE;
 
     caps.dwSize = sizeof(caps);
     hr = object->GetCaps( &caps );
@@ -4692,7 +4732,7 @@ static bool CALLBACK deviceCountCallback( LPGUID lpguid,
     DSCAPS caps;
     LPDIRECTSOUND object;
     hr = DirectSoundCreate(  lpguid, &object,   NULL );
-    if ( hr != DS_OK ) return true;
+    if ( hr != DS_OK ) return TRUE;
 
     caps.dwSize = sizeof(caps);
     hr = object->GetCaps( &caps );
@@ -4703,20 +4743,20 @@ static bool CALLBACK deviceCountCallback( LPGUID lpguid,
     object->Release();
   }
 
-  if ( info->getDefault && lpguid == NULL ) return false;
+  if ( info->getDefault && lpguid == NULL ) return FALSE;
 
   if ( info->findIndex && info->counter > info->index ) {
     info->id = lpguid;
     info->name = convertTChar( description );
-    return false;
+    return FALSE;
   }
 
-  return true;
+  return TRUE;
 }
 
 static char* getErrorString( int code )
 {
-	switch (code) {
+	switch ( code ) {
 
   case DSERR_ALLOCATED:
     return "Already allocated";
@@ -5906,7 +5946,9 @@ extern "C" void *alsaCallbackHandler( void *ptr )
 #ifdef SCHED_RR
   // Set a higher scheduler priority (P.J. Leonard)
   struct sched_param param;
-  param.sched_priority = 39;   // Is this the best number?
+  int min = sched_get_priority_min( SCHED_RR );
+  int max = sched_get_priority_max( SCHED_RR );
+  param.sched_priority = min + ( max - min ) / 2;   // Is this the best number?
   sched_setscheduler( 0, SCHED_RR, &param );
 #endif
 
@@ -5928,7 +5970,7 @@ extern "C" void *alsaCallbackHandler( void *ptr )
 #include <sys/ioctl.h>
 #include <unistd.h>
 #include <fcntl.h>
-#include "oss/soundcard.h"
+#include "soundcard.h"
 #include <errno.h>
 #include <math.h>
 

src/RtMidi.cpp

@@ -8,7 +8,7 @@
     RtMidi WWW site: http://music.mcgill.ca/~gary/rtmidi/
 
     RtMidi: realtime MIDI i/o C++ classes
-    Copyright (c) 2003-2005 Gary P. Scavone
+    Copyright (c) 2003-2007 Gary P. Scavone
 
     Permission is hereby granted, free of charge, to any person
     obtaining a copy of this software and associated documentation files
@@ -35,7 +35,7 @@
 */
 /**********************************************************************/
 
-// RtMidi: Version 1.0.5, in development
+// RtMidi: Version 1.0.7
 
 #include "RtMidi.h"
 #include <sstream>
@@ -768,12 +768,9 @@ extern "C" void *alsaMidiHandler( void *ptr )
       else
         message.bytes.insert( message.bytes.end(), buffer, &buffer[nBytes] );
 
-      if ( ev->type == SND_SEQ_EVENT_SYSEX && message.bytes.back() == 0xF7 )
-        continueSysex = false;
-      else {
-        continueSysex = true;
+      continueSysex = ( ev->type == SND_SEQ_EVENT_SYSEX && message.bytes.back() != 0xF7 );
+      if ( continueSysex )
         break;
-      }
 
       // Calculate the time stamp:
       message.timeStamp = 0.0;
@@ -1926,7 +1923,7 @@ std::string RtMidiIn :: getPortName( unsigned int portNumber )
   }
 
   MIDIINCAPS deviceCaps;
-  MMRESULT result = midiInGetDevCaps( portNumber, &deviceCaps, sizeof(MIDIINCAPS));
+  midiInGetDevCaps( portNumber, &deviceCaps, sizeof(MIDIINCAPS));
 
   // For some reason, we need to copy character by character with
   // UNICODE (thanks to Eduardo Coutinho!).
@@ -1960,7 +1957,7 @@ std::string RtMidiOut :: getPortName( unsigned int portNumber )
   }
 
   MIDIOUTCAPS deviceCaps;
-  MMRESULT result = midiOutGetDevCaps( portNumber, &deviceCaps, sizeof(MIDIOUTCAPS));
+  midiOutGetDevCaps( portNumber, &deviceCaps, sizeof(MIDIOUTCAPS));
 
   // For some reason, we need to copy character by character with
   // UNICODE (thanks to Eduardo Coutinho!).

src/RtWvOut.cpp

@@ -104,7 +104,7 @@ RtWvOut :: RtWvOut( unsigned int nChannels, StkFloat sampleRate, int device, int
   }
 
   data_.resize( size * nBuffers, nChannels );
-  unsigned int offset = (unsigned int ) (data_.size() / 2.0);
+  unsigned int offset = (unsigned int ) (data_.frames() / 2.0);
   writeIndex_ = offset;    // start writing half-way into buffer
   framesFilled_ = offset;
 }

src/Stk.cpp

@@ -8,7 +8,21 @@
     provides error handling and byte-swapping
     functions.
 
-    by Perry R. Cook and Gary P. Scavone, 1995 - 2007.
+    The Synthesis ToolKit in C++ (STK) is a set of open source audio
+    signal processing and algorithmic synthesis classes written in the
+    C++ programming language. STK was designed to facilitate rapid
+    development of music synthesis and audio processing software, with
+    an emphasis on cross-platform functionality, realtime control,
+    ease of use, and educational example code.  STK currently runs
+    with realtime support (audio and MIDI) on Linux, Macintosh OS X,
+    and Windows computer platforms. Generic, non-realtime support has
+    been tested under NeXTStep, Sun, and other platforms and should
+    work with any standard C++ compiler.
+
+    STK WWW site: http://ccrma.stanford.edu/software/stk/
+
+    The Synthesis ToolKit in C++ (STK)
+    Copyright (c) 1995-2007 Perry R. Cook and Gary P. Scavone
 
     Permission is hereby granted, free of charge, to any person
     obtaining a copy of this software and associated documentation files