Updates to eguitar, ragamatic and examples project files for new RtAudio API.

projects/demo/demo.cpp

@@ -203,7 +203,6 @@ int main( int argc, char *argv[] )
   int i;
 
 #if defined(__STK_REALTIME__)
-  //RtAudio dac( RtAudio::UNSPECIFIED );
   RtAudio *dac = 0;
 #endif
 

projects/eguitar/eguitar.cpp

@@ -265,7 +265,7 @@ int main( int argc, char *argv[] )
   int i;
 
 #if defined(__STK_REALTIME__)
-  RtAudio dac;
+  RtAudio *dac = 0;
 #endif
 
   // If you want to change the default sample rate (set in Stk.h), do
@@ -294,16 +294,14 @@ int main( int argc, char *argv[] )
   // If realtime output, allocate the dac here.
 #if defined(__STK_REALTIME__)
   if ( data.realtime ) {
+    dac = (RtAudio *) new RtAudio( RtAudio::UNSPECIFIED );
     RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
     RtAudio::StreamParameters parameters;
-    parameters.deviceId = dac.getDefaultOutputDevice();
+    parameters.deviceId = dac->getDefaultOutputDevice();
     parameters.nChannels = data.channels;
     unsigned int bufferFrames = RT_BUFFER_SIZE;
-    try {
+    if ( dac->openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data ) ) {
-      dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data );
+      std::cout << dac->getErrorText() << std::endl;
-    }
-    catch ( RtAudioError& error ) {
-      error.printMessage();
       goto cleanup;
     }
   }
@@ -335,11 +333,8 @@ int main( int argc, char *argv[] )
   // If realtime output, set our callback function and start the dac.
 #if defined(__STK_REALTIME__)
   if ( data.realtime ) {
-    try {
+    if ( dac->startStream() ) {
-      dac.startStream();
+      std::cout << dac->getErrorText() << std::endl;
-    }
-    catch ( RtAudioError &error ) {
-      error.printMessage();
       goto cleanup;
     }
   }
@@ -359,14 +354,8 @@ int main( int argc, char *argv[] )
 
   // Shut down the output stream.
 #if defined(__STK_REALTIME__)
-  if ( data.realtime ) {
+  if ( data.realtime )
-    try {
+    dac->closeStream();
-      dac.closeStream();
-    }
-    catch ( RtAudioError& error ) {
-      error.printMessage();
-    }
-  }
 #endif
 
  cleanup:
@@ -374,6 +363,7 @@ int main( int argc, char *argv[] )
   for ( i=0; i<(int)data.nWvOuts; i++ ) delete data.wvout[i];
   free( data.wvout );
   delete data.guitar;
+  delete dac;
 
 	std::cout << "\nStk eguitar finished ... goodbye.\n\n";
   return 0;

projects/examples/audioprobe.cpp

@@ -11,76 +11,107 @@
 #include <iostream>
 #include <map>
 
-int main()
+void usage( void ) {
-{
+  // Error function in case of incorrect command-line
-  // Create an api map.
+  // argument specifications
-  std::map<int, std::string> apiMap;
+  std::cout << "\nuseage: audioprobe <apiname> <nRepeats>\n";
-  apiMap[RtAudio::MACOSX_CORE] = "OS-X CoreAudio";
+  std::cout << "    where apiname = an optional api (ex., 'core', default = all compiled),\n";
-  apiMap[RtAudio::WINDOWS_ASIO] = "Windows ASIO";
+  std::cout << "    and nRepeats = an optional number of times to repeat the device query (default = 0),\n";
-  apiMap[RtAudio::WINDOWS_DS] = "Windows DirectSound";
+  std::cout << "                   which can be used to test device (dis)connections.\n\n";
-  apiMap[RtAudio::UNIX_JACK] = "Jack Client";
+  exit( 0 );
-  apiMap[RtAudio::LINUX_ALSA] = "Linux ALSA";
+}
-  apiMap[RtAudio::LINUX_OSS] = "Linux OSS";
-  apiMap[RtAudio::RTAUDIO_DUMMY] = "RtAudio Dummy";
 
+std::vector< RtAudio::Api > listApis()
+{
   std::vector< RtAudio::Api > apis;
   RtAudio :: getCompiledApi( apis );
 
   std::cout << "\nCompiled APIs:\n";
-  for ( unsigned int i=0; i<apis.size(); i++ )
+  for ( size_t i=0; i<apis.size(); i++ )
-    std::cout << "  " << apiMap[ apis[i] ] << std::endl;
+    std::cout << i << ". " << RtAudio::getApiDisplayName(apis[i])
+              << " (" << RtAudio::getApiName(apis[i]) << ")" << std::endl;
 
-  RtAudio audio;
+  return apis;
+}
+
+void listDevices( RtAudio& audio )
+{
   RtAudio::DeviceInfo info;
 
-  std::cout << "\nCurrent API: " << apiMap[ audio.getCurrentApi() ] << std::endl;
+  std::cout << "\nAPI: " << RtAudio::getApiDisplayName(audio.getCurrentApi()) << std::endl;
 
-  unsigned int devices = audio.getDeviceCount();
+  std::vector<unsigned int> devices = audio.getDeviceIds();
-  std::cout << "\nFound " << devices << " device(s) ...\n";
+  std::cout << "\nFound " << devices.size() << " device(s) ...\n";
 
-  for (unsigned int i=0; i<devices; i++) {
+  for (unsigned int i=0; i<devices.size(); i++) {
-    info = audio.getDeviceInfo(i);
+    info = audio.getDeviceInfo( devices[i] );
 
     std::cout << "\nDevice Name = " << info.name << '\n';
-    if ( info.probed == false )
+    std::cout << "Device Index = " << i << '\n';
-      std::cout << "Probe Status = UNsuccessful\n";
+    std::cout << "Output Channels = " << info.outputChannels << '\n';
+    std::cout << "Input Channels = " << info.inputChannels << '\n';
+    std::cout << "Duplex Channels = " << info.duplexChannels << '\n';
+    if ( info.isDefaultOutput ) std::cout << "This is the default output device.\n";
+    else std::cout << "This is NOT the default output device.\n";
+    if ( info.isDefaultInput ) std::cout << "This is the default input device.\n";
+    else std::cout << "This is NOT the default input device.\n";
+    if ( info.nativeFormats == 0 )
+      std::cout << "No natively supported data formats(?)!";
     else {
-      std::cout << "Probe Status = Successful\n";
+      std::cout << "Natively supported data formats:\n";
-      std::cout << "Output Channels = " << info.outputChannels << '\n';
+      if ( info.nativeFormats & RTAUDIO_SINT8 )
-      std::cout << "Input Channels = " << info.inputChannels << '\n';
+        std::cout << "  8-bit int\n";
-      std::cout << "Duplex Channels = " << info.duplexChannels << '\n';
+      if ( info.nativeFormats & RTAUDIO_SINT16 )
-      if ( info.isDefaultOutput ) std::cout << "This is the default output device.\n";
+        std::cout << "  16-bit int\n";
-      else std::cout << "This is NOT the default output device.\n";
+      if ( info.nativeFormats & RTAUDIO_SINT24 )
-      if ( info.isDefaultInput ) std::cout << "This is the default input device.\n";
+        std::cout << "  24-bit int\n";
-      else std::cout << "This is NOT the default input device.\n";
+      if ( info.nativeFormats & RTAUDIO_SINT32 )
-      if ( info.nativeFormats == 0 )
+        std::cout << "  32-bit int\n";
-        std::cout << "No natively supported data formats(?)!";
+      if ( info.nativeFormats & RTAUDIO_FLOAT32 )
-      else {
+        std::cout << "  32-bit float\n";
-        std::cout << "Natively supported data formats:\n";
+      if ( info.nativeFormats & RTAUDIO_FLOAT64 )
-        if ( info.nativeFormats & RTAUDIO_SINT8 )
+        std::cout << "  64-bit float\n";
-          std::cout << "  8-bit int\n";
+    }
-        if ( info.nativeFormats & RTAUDIO_SINT16 )
+    if ( info.sampleRates.size() < 1 )
-          std::cout << "  16-bit int\n";
+      std::cout << "No supported sample rates found!";
-        if ( info.nativeFormats & RTAUDIO_SINT24 )
+    else {
-          std::cout << "  24-bit int\n";
+      std::cout << "Supported sample rates = ";
-        if ( info.nativeFormats & RTAUDIO_SINT32 )
+      for (unsigned int j=0; j<info.sampleRates.size(); j++)
-          std::cout << "  32-bit int\n";
+        std::cout << info.sampleRates[j] << " ";
-        if ( info.nativeFormats & RTAUDIO_FLOAT32 )
+    }
-          std::cout << "  32-bit float\n";
+    std::cout << std::endl;
-        if ( info.nativeFormats & RTAUDIO_FLOAT64 )
+    if ( info.preferredSampleRate == 0 )
-          std::cout << "  64-bit float\n";
+      std::cout << "No preferred sample rate found!" << std::endl;
+    else
+      std::cout << "Preferred sample rate = " << info.preferredSampleRate << std::endl;
+  }
+}
+
+int main(int argc, char *argv[])
+{
+  std::cout << "\nRtAudio Version " << RtAudio::getVersion() << std::endl;
+
+  std::vector< RtAudio::Api > apis = listApis();
+
+  // minimal command-line checking
+  if (argc > 3 ) usage();
+  unsigned int nRepeats = 0;
+  if ( argc > 2 ) nRepeats = (unsigned int) atoi( argv[2] );
+
+  char input;
+  for ( size_t api=0; api < apis.size(); api++ ) {
+    if (argc < 2 || apis[api] == RtAudio::getCompiledApiByName(argv[1]) ) {
+      RtAudio audio(apis[api]);
+      for ( size_t n=0; n <= nRepeats; n++ ) {
+        listDevices(audio);
+        if ( n < nRepeats ) {
+          std::cout << std::endl;
+          std::cout << "\nWaiting ... press <enter> to repeat.\n";
+          std::cin.get(input);
+        }
       }
-      if ( info.sampleRates.size() < 1 )
-        std::cout << "No supported sample rates found!";
-      else {
-        std::cout << "Supported sample rates = ";
-        for (unsigned int j=0; j<info.sampleRates.size(); j++)
-          std::cout << info.sampleRates[j] << " ";
-      }
-      std::cout << std::endl;
     }
   }
-  std::cout << std::endl;
 
   return 0;
 }

projects/examples/bethree.cpp

@@ -56,11 +56,8 @@ int main()
   parameters.nChannels = 1;
   RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
   unsigned int bufferFrames = RT_BUFFER_SIZE;
-  try {
+  if ( dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data ) ) {
-    dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data );
+    std::cout << dac.getErrorText() << std::endl;
-  }
-  catch ( RtAudioError& error ) {
-    error.printMessage();
     goto cleanup;
   }
 
@@ -75,11 +72,8 @@ int main()
   data.frequency = 220.0;
   data.instrument->noteOn( data.frequency, 0.5 );
 
-  try {
+  if ( dac.startStream() ) {
-    dac.startStream();
+    std::cout << dac.getErrorText() << std::endl;
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
     goto cleanup;
   }
 
@@ -88,12 +82,7 @@ int main()
     Stk::sleep( 100 );
   
   // Shut down the callback and output stream.
-  try {
+  dac.closeStream();
-    dac.closeStream();
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
-  }
 
  cleanup:
   delete data.instrument;

projects/examples/controlbee.cpp

@@ -130,11 +130,8 @@ int main( int argc, char *argv[] )
   parameters.nChannels = 1;
   RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
   unsigned int bufferFrames = RT_BUFFER_SIZE;
-  try {
+  if ( dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data ) ) {
-    dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data );
+    std::cout << dac.getErrorText() << std::endl;
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
     goto cleanup;
   }
 
@@ -149,11 +146,8 @@ int main( int argc, char *argv[] )
   if ( data.messager.setScoreFile( argv[1] ) == false )
     goto cleanup;
 
-  try {
+  if ( dac.startStream() ) {
-    dac.startStream();
+    std::cout << dac.getErrorText() << std::endl;
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
     goto cleanup;
   }
 
@@ -162,12 +156,7 @@ int main( int argc, char *argv[] )
     Stk::sleep( 100 );
   
   // Shut down the output stream.
-  try {
+  dac.closeStream();
-    dac.closeStream();
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
-  }
 
  cleanup:
   delete data.instrument;

projects/examples/crtsine.cpp

@@ -33,21 +33,15 @@ int main()
   parameters.nChannels = 1;
   RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
   unsigned int bufferFrames = RT_BUFFER_SIZE;
-  try {
+  if ( dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&sine ) ) {
-    dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&sine );
+    std::cout << dac.getErrorText() << std::endl;
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
     goto cleanup;
   }
 
   sine.setFrequency(440.0);
 
-  try {
+  if ( dac.startStream() ) {
-    dac.startStream();
+    std::cout << dac.getErrorText() << std::endl;
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
     goto cleanup;
   }
 
@@ -57,12 +51,7 @@ int main()
   std::cin.get( keyhit );
 
   // Shut down the output stream.
-  try {
+  dac.closeStream();
-    dac.closeStream();
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
-  }
 
  cleanup:
 

projects/examples/duplex.cpp

@@ -1,7 +1,7 @@
 /******************************************/
 /*
   duplex.cpp
-  by Gary P. Scavone, 2006-2007.
+  by Gary P. Scavone, 2006-2019.
 
   This program opens a duplex stream and passes
   input directly through to the output.
@@ -14,24 +14,26 @@
 #include <cstring>
 
 /*
-typedef signed long  MY_TYPE;
+typedef char MY_TYPE;
-#define FORMAT RTAUDIO_SINT24
-
-typedef char  MY_TYPE;
 #define FORMAT RTAUDIO_SINT8
-
-typedef signed short  MY_TYPE;
-#define FORMAT RTAUDIO_SINT16
-
-typedef signed long  MY_TYPE;
-#define FORMAT RTAUDIO_SINT32
-
-typedef float  MY_TYPE;
-#define FORMAT RTAUDIO_FLOAT32
 */
 
-typedef double  MY_TYPE;
+typedef signed short MY_TYPE;
+#define FORMAT RTAUDIO_SINT16
+
+/*
+typedef S24 MY_TYPE;
+#define FORMAT RTAUDIO_SINT24
+
+typedef signed long MY_TYPE;
+#define FORMAT RTAUDIO_SINT32
+
+typedef float MY_TYPE;
+#define FORMAT RTAUDIO_FLOAT32
+
+typedef double MY_TYPE;
 #define FORMAT RTAUDIO_FLOAT64
+*/
 
 void usage( void ) {
   // Error function in case of incorrect command-line
@@ -39,26 +41,52 @@ void usage( void ) {
   std::cout << "\nuseage: duplex N fs <iDevice> <oDevice> <iChannelOffset> <oChannelOffset>\n";
   std::cout << "    where N = number of channels,\n";
   std::cout << "    fs = the sample rate,\n";
-  std::cout << "    iDevice = optional input device to use (default = 0),\n";
+  std::cout << "    iDevice = optional input device index to use (default = 0),\n";
-  std::cout << "    oDevice = optional output device to use (default = 0),\n";
+  std::cout << "    oDevice = optional output device index to use (default = 0),\n";
   std::cout << "    iChannelOffset = an optional input channel offset (default = 0),\n";
   std::cout << "    and oChannelOffset = optional output channel offset (default = 0).\n\n";
   exit( 0 );
 }
 
-int inout( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
+unsigned int getDeviceIndex( std::vector<std::string> deviceNames, bool isInput = false )
+{
+  unsigned int i;
+  std::string keyHit;
+  std::cout << '\n';
+  for ( i=0; i<deviceNames.size(); i++ )
+    std::cout << "  Device #" << i << ": " << deviceNames[i] << '\n';
+  do {
+    if ( isInput )
+      std::cout << "\nChoose an input device #: ";
+    else
+      std::cout << "\nChoose an output device #: ";
+    std::cin >> i;
+  } while ( i >= deviceNames.size() );
+  std::getline( std::cin, keyHit );  // used to clear out stdin
+  return i;
+}
+
+double streamTimePrintIncrement = 1.0; // seconds
+double streamTimePrintTime = 1.0; // seconds
+
+int inout( void *outputBuffer, void *inputBuffer, unsigned int /*nBufferFrames*/,
            double streamTime, RtAudioStreamStatus status, void *data )
 {
   // Since the number of input and output channels is equal, we can do
   // a simple buffer copy operation here.
   if ( status ) std::cout << "Stream over/underflow detected." << std::endl;
 
-  unsigned long *bytes = (unsigned long *) data;
+  if ( streamTime >= streamTimePrintTime ) {
+    std::cout << "streamTime = " << streamTime << std::endl;
+    streamTimePrintTime += streamTimePrintIncrement;
+  }
+
+  unsigned int *bytes = (unsigned int *) data;
   memcpy( outputBuffer, inputBuffer, *bytes );
   return 0;
 }
 
-int main(int argc, char *argv[])
+int main( int argc, char *argv[] )
 {
   unsigned int channels, fs, bufferBytes, oDevice = 0, iDevice = 0, iOffset = 0, oOffset = 0;
 
@@ -66,9 +94,10 @@ int main(int argc, char *argv[])
   if (argc < 3 || argc > 7 ) usage();
 
   RtAudio adac;
-  if ( adac.getDeviceCount() < 1 ) {
+  std::vector<unsigned int> deviceIds = adac.getDeviceIds();
+  if ( deviceIds.size() < 1 ) {
     std::cout << "\nNo audio devices found!\n";
-    exit( 0 );
+    exit( 1 );
   }
 
   channels = (unsigned int) atoi(argv[1]);
@@ -88,48 +117,48 @@ int main(int argc, char *argv[])
   // Set the same number of channels for both input and output.
   unsigned int bufferFrames = 512;
   RtAudio::StreamParameters iParams, oParams;
-  if ( iDevice == 0 )
-    iParams.deviceId = adac.getDefaultInputDevice();
-  else
-    iParams.deviceId = iDevice - 1;
   iParams.nChannels = channels;
   iParams.firstChannel = iOffset;
-  if ( oDevice == 0 )
-    oParams.deviceId = adac.getDefaultOutputDevice();
-  else
-    oParams.deviceId = oDevice - 1;
   oParams.nChannels = channels;
   oParams.firstChannel = oOffset;
 
+  if ( iDevice == 0 )
+    iParams.deviceId = adac.getDefaultInputDevice();
+  else {
+    if ( iDevice >= deviceIds.size() )
+      iDevice = getDeviceIndex( adac.getDeviceNames(), true );
+    iParams.deviceId = deviceIds[iDevice];
+  }
+  if ( oDevice == 0 )
+    oParams.deviceId = adac.getDefaultOutputDevice();
+  else {
+    if ( oDevice >= deviceIds.size() )
+      oDevice = getDeviceIndex( adac.getDeviceNames() );
+    oParams.deviceId = deviceIds[oDevice];
+  }
+  
   RtAudio::StreamOptions options;
   //options.flags |= RTAUDIO_NONINTERLEAVED;
 
   bufferBytes = bufferFrames * channels * sizeof( MY_TYPE );
-  try {
+  if ( adac.openStream( &oParams, &iParams, FORMAT, fs, &bufferFrames, &inout, (void *)&bufferBytes, &options ) ) {
-    adac.openStream( &oParams, &iParams, FORMAT, fs, &bufferFrames, &inout, (void *)&bufferBytes, &options );
+    goto cleanup;
-  }
-  catch ( RtAudioError& e ) {
-    std::cout << '\n' << e.getMessage() << '\n' << std::endl;
-    exit( 1 );
   }
 
+  if ( adac.isStreamOpen() == false ) goto cleanup;
+
   // Test RtAudio functionality for reporting latency.
   std::cout << "\nStream latency = " << adac.getStreamLatency() << " frames" << std::endl;
 
-  try {
+  if ( adac.startStream() ) goto cleanup;
-    adac.startStream();
 
-    char input;
+  char input;
-    std::cout << "\nRunning ... press <enter> to quit (buffer frames = " << bufferFrames << ").\n";
+  std::cout << "\nRunning ... press <enter> to quit (buffer frames = " << bufferFrames << ").\n";
-    std::cin.get(input);
+  std::cin.get(input);
 
-    // Stop the stream.
+  // Stop the stream.
+  if ( adac.isStreamRunning() )
     adac.stopStream();
-  }
-  catch ( RtAudioError& e ) {
-    std::cout << '\n' << e.getMessage() << '\n' << std::endl;
-    goto cleanup;
-  }
 
  cleanup:
   if ( adac.isStreamOpen() ) adac.closeStream();

projects/examples/grains.cpp

@@ -79,35 +79,23 @@ int main( int argc, char *argv[] )
   parameters.nChannels = grani.channelsOut();
   RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
   unsigned int bufferFrames = RT_BUFFER_SIZE;
-  try {
+  if ( dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&grani ) ) {
-    dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&grani );
+    std::cout << dac.getErrorText() << std::endl;
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
     goto cleanup;
   }
 
-  try {
+  if ( dac.startStream() ) {
-    dac.startStream();
+    std::cout << dac.getErrorText() << std::endl;
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
     goto cleanup;
   }
 
-
   // Block waiting here.
   char keyhit;
   std::cout << "\nPlaying ... press <enter> to quit.\n";
   std::cin.get( keyhit );
 
   // Shut down the callback and output stream.
-  try {
+  dac.closeStream();
-    dac.closeStream();
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
-  }
 
  cleanup:
 

projects/examples/play.cpp

@@ -99,11 +99,8 @@ int main(int argc, char *argv[])
   parameters.nChannels = ( channels == 1 ) ? 2 : channels; //  Play mono files as stereo.
   RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
   unsigned int bufferFrames = RT_BUFFER_SIZE;
-  try {
+  if ( dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&input ) ) {
-    dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&input );
+    std::cout << dac.getErrorText() << std::endl;
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
     goto cleanup;
   }
 
@@ -113,11 +110,8 @@ int main(int argc, char *argv[])
   // Resize the StkFrames object appropriately.
   frames.resize( bufferFrames, channels );
 
-  try {
+  if ( dac.startStream() ) {
-    dac.startStream();
+    std::cout << dac.getErrorText() << std::endl;
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
     goto cleanup;
   }
 
@@ -127,12 +121,7 @@ int main(int argc, char *argv[])
   
   // By returning a non-zero value in the callback above, the stream
   // is automatically stopped.  But we should still close it.
-  try {
+  dac.closeStream();
-    dac.closeStream();
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
-  }
 
  cleanup:
   return 0;

projects/examples/threebees.cpp

@@ -130,11 +130,8 @@ int main()
   parameters.nChannels = 1;
   RtAudioFormat format = ( sizeof(StkFloat) == 8 ) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
   unsigned int bufferFrames = RT_BUFFER_SIZE;
-  try {
+  if ( dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data ) ) {
-    dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data );
+    std::cout << dac.getErrorText() << std::endl;
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
     goto cleanup;
   }
 
@@ -154,11 +151,8 @@ int main()
   if ( data.messager.startStdInput() == false )
     goto cleanup;
 
-  try {
+  if ( dac.startStream() ) {
-    dac.startStream();
+    std::cout << dac.getErrorText() << std::endl;
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
     goto cleanup;
   }
 
@@ -167,12 +161,7 @@ int main()
     Stk::sleep( 100 );
   
   // Shut down the callback and output stream.
-  try {
+  dac.closeStream();
-    dac.closeStream();
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
-  }
 
  cleanup:
   for ( i=0; i<3; i++ ) delete instrument[i];

projects/ragamatic/ragamat.cpp

@@ -291,11 +291,8 @@ int main( int argc, char *argv[] )
   parameters.deviceId = dac.getDefaultOutputDevice();
   parameters.nChannels = 2;
   unsigned int bufferFrames = RT_BUFFER_SIZE;
-  try {
+  if ( dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data ) ) {
-    dac.openStream( &parameters, NULL, format, (unsigned int)Stk::sampleRate(), &bufferFrames, &tick, (void *)&data );
+    std::cout << dac.getErrorText() << std::endl;
-  }
-  catch ( RtAudioError& error ) {
-    error.printMessage();
     goto cleanup;
   }
 
@@ -314,11 +311,8 @@ int main( int argc, char *argv[] )
 	(void) signal( SIGINT, finish );
 
   // If realtime output, set our callback function and start the dac.
-  try {
+  if ( dac.startStream() ) {
-    dac.startStream();
+    std::cout << dac.getErrorText() << std::endl;
-  }
-  catch ( RtAudioError &error ) {
-    error.printMessage();
     goto cleanup;
   }
 
@@ -329,15 +323,9 @@ int main( int argc, char *argv[] )
   }
 
   // Shut down the output stream.
-  try {
+  dac.closeStream();
-    dac.closeStream();
-  }
-  catch ( RtAudioError& error ) {
-    error.printMessage();
-  }
 
  cleanup:
 
   return 0;
-
 }