Merge pull request #67 from antis81/refactor/cpp0x-compatibility

C++0x compatible exception semantics: remove throw()

diff --git a/CMakeLists.txt b/CMakeLists.txt
index 9de6a85dada99036a3020fd3ac742359a2af4370..9117e40c368fac612429f5264ecebf93ff77f601 100644 (file)
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -44,7 +44,8 @@ elseif (UNIX AND NOT APPLE)
     if (AUDIO_LINUX_PULSE)
         find_library(PULSE_LIB pulse)
         find_library(PULSESIMPLE_LIB pulse-simple)
-        list(APPEND LINKLIBS ${PULSE_LIB} ${PULSESIMPLE_LIB})
+        find_package(Threads REQUIRED CMAKE_THREAD_PREFER_PTHREAD)
+        list(APPEND LINKLIBS ${PULSE_LIB} ${PULSESIMPLE_LIB} ${CMAKE_THREAD_LIBS_INIT})
         add_definitions(-D__LINUX_PULSE__)
         message(STATUS "Using Linux PulseAudio")
     endif (AUDIO_LINUX_PULSE)

diff --git a/RtAudio.cpp b/RtAudio.cpp
index 8fa49a3bc8fe0ba848cc59331c20e98de62109b5..91bd22165290c13c919f8b26d5de65fb9273877d 100644 (file)
--- a/RtAudio.cpp
+++ b/RtAudio.cpp
@@ -1,4 +1,4 @@
-/************************************************************************/\r
+/************************************************************************/\r
 /*! \class RtAudio\r
     \brief Realtime audio i/o C++ classes.\r
 \r
@@ -3859,8 +3859,7 @@ private:
 // In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate\r
 // between HW and the user. The convertBufferWasapi function is used to perform this conversion\r
 // between HwIn->UserIn and UserOut->HwOut during the stream callback loop.\r
-// This sample rate converter favors speed over quality, and works best with conversions between\r
-// one rate and its multiple.\r
+// This sample rate converter works best with conversions between one rate and its multiple.\r
 void convertBufferWasapi( char* outBuffer,\r
                           const char* inBuffer,\r
                           const unsigned int& channelCount,\r
@@ -3872,40 +3871,129 @@ void convertBufferWasapi( char* outBuffer,
 {\r
   // calculate the new outSampleCount and relative sampleStep\r
   float sampleRatio = ( float ) outSampleRate / inSampleRate;\r
+  float sampleRatioInv = ( float ) 1 / sampleRatio;\r
   float sampleStep = 1.0f / sampleRatio;\r
   float inSampleFraction = 0.0f;\r
 \r
   outSampleCount = ( unsigned int ) roundf( inSampleCount * sampleRatio );\r
 \r
-  // frame-by-frame, copy each relative input sample into it's corresponding output sample\r
-  for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )\r
+  // if inSampleRate is a multiple of outSampleRate (or vice versa) there's no need to interpolate\r
+  if ( floor( sampleRatio ) == sampleRatio || floor( sampleRatioInv ) == sampleRatioInv )\r
   {\r
-    unsigned int inSample = ( unsigned int ) inSampleFraction;\r
-\r
-    switch ( format )\r
+    // frame-by-frame, copy each relative input sample into it's corresponding output sample\r
+    for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )\r
     {\r
-      case RTAUDIO_SINT8:\r
-        memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );\r
-        break;\r
-      case RTAUDIO_SINT16:\r
-        memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );\r
-        break;\r
-      case RTAUDIO_SINT24:\r
-        memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );\r
-        break;\r
-      case RTAUDIO_SINT32:\r
-        memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );\r
-        break;\r
-      case RTAUDIO_FLOAT32:\r
-        memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );\r
-        break;\r
-      case RTAUDIO_FLOAT64:\r
-        memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );\r
-        break;\r
+      unsigned int inSample = ( unsigned int ) inSampleFraction;\r
+\r
+      switch ( format )\r
+      {\r
+        case RTAUDIO_SINT8:\r
+          memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );\r
+          break;\r
+        case RTAUDIO_SINT16:\r
+          memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );\r
+          break;\r
+        case RTAUDIO_SINT24:\r
+          memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );\r
+          break;\r
+        case RTAUDIO_SINT32:\r
+          memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );\r
+          break;\r
+        case RTAUDIO_FLOAT32:\r
+          memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );\r
+          break;\r
+        case RTAUDIO_FLOAT64:\r
+          memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );\r
+          break;\r
+      }\r
+\r
+      // jump to next in sample\r
+      inSampleFraction += sampleStep;\r
     }\r
+  }\r
+  else // else interpolate\r
+  {\r
+    // frame-by-frame, copy each relative input sample into it's corresponding output sample\r
+    for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )\r
+    {\r
+      unsigned int inSample = ( unsigned int ) inSampleFraction;\r
+      float inSampleDec = inSampleFraction - inSample;\r
+      unsigned int frameInSample = inSample * channelCount;\r
+      unsigned int frameOutSample = outSample * channelCount;\r
+\r
+      switch ( format )\r
+      {\r
+        case RTAUDIO_SINT8:\r
+        {\r
+          for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+          {\r
+            char fromSample = ( ( char* ) inBuffer )[ frameInSample + channel ];\r
+            char toSample = ( ( char* ) inBuffer )[ frameInSample + channelCount + channel ];\r
+            char sampleDiff = ( char ) ( ( toSample - fromSample ) * inSampleDec );\r
+            ( ( char* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+          }\r
+          break;\r
+        }\r
+        case RTAUDIO_SINT16:\r
+        {\r
+          for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+          {\r
+            short fromSample = ( ( short* ) inBuffer )[ frameInSample + channel ];\r
+            short toSample = ( ( short* ) inBuffer )[ frameInSample + channelCount + channel ];\r
+            short sampleDiff = ( short ) ( ( toSample - fromSample ) * inSampleDec );\r
+            ( ( short* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+          }\r
+          break;\r
+        }\r
+        case RTAUDIO_SINT24:\r
+        {\r
+          for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+          {\r
+            int fromSample = ( ( S24* ) inBuffer )[ frameInSample + channel ].asInt();\r
+            int toSample = ( ( S24* ) inBuffer )[ frameInSample + channelCount + channel ].asInt();\r
+            int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );\r
+            ( ( S24* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+          }\r
+          break;\r
+        }\r
+        case RTAUDIO_SINT32:\r
+        {\r
+          for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+          {\r
+            int fromSample = ( ( int* ) inBuffer )[ frameInSample + channel ];\r
+            int toSample = ( ( int* ) inBuffer )[ frameInSample + channelCount + channel ];\r
+            int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );\r
+            ( ( int* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+          }\r
+          break;\r
+        }\r
+        case RTAUDIO_FLOAT32:\r
+        {\r
+          for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+          {\r
+            float fromSample = ( ( float* ) inBuffer )[ frameInSample + channel ];\r
+            float toSample = ( ( float* ) inBuffer )[ frameInSample + channelCount + channel ];\r
+            float sampleDiff = ( toSample - fromSample ) * inSampleDec;\r
+            ( ( float* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+          }\r
+          break;\r
+        }\r
+        case RTAUDIO_FLOAT64:\r
+        {\r
+          for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+          {\r
+            double fromSample = ( ( double* ) inBuffer )[ frameInSample + channel ];\r
+            double toSample = ( ( double* ) inBuffer )[ frameInSample + channelCount + channel ];\r
+            double sampleDiff = ( toSample - fromSample ) * inSampleDec;\r
+            ( ( double* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+          }\r
+          break;\r
+        }\r
+      }\r
 \r
-    // jump to next in sample\r
-    inSampleFraction += sampleStep;\r
+      // jump to next in sample\r
+      inSampleFraction += sampleStep;\r
+    }\r
   }\r
 }\r
 \r
@@ -8695,8 +8783,10 @@ RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
     info.nativeFormats |= RTAUDIO_SINT8;\r
   if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )\r
     info.nativeFormats |= RTAUDIO_SINT32;\r
+#ifdef AFMT_FLOAT\r
   if ( mask & AFMT_FLOAT )\r
     info.nativeFormats |= RTAUDIO_FLOAT32;\r
+#endif\r
   if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )\r
     info.nativeFormats |= RTAUDIO_SINT24;\r
 \r
@@ -9023,7 +9113,7 @@ bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned
   }\r
 \r
   // Verify the sample rate setup worked.\r
-  if ( abs( srate - sampleRate ) > 100 ) {\r
+  if ( abs( srate - (int)sampleRate ) > 100 ) {\r
     close( fd );\r
     errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";\r
     errorText_ = errorStream_.str();\r

diff --git a/RtAudio.h b/RtAudio.h
index d7fefa3e53a2ab3d50309f7679c343e8ebce606a..f9d56ff6490c8e3d00255d4eb430b66c9110f917 100644 (file)
--- a/RtAudio.h
+++ b/RtAudio.h
@@ -616,7 +616,7 @@ struct CallbackInfo {
 
   // Default constructor.
   CallbackInfo()
-  :object(0), callback(0), userData(0), errorCallback(0), apiInfo(0), isRunning(false), doRealtime(false) {}
+  :object(0), callback(0), userData(0), errorCallback(0), apiInfo(0), isRunning(false), doRealtime(false), priority(0) {}
 };
 
 // **************************************************************** //

diff --git a/contrib/python/pyrtaudio/rtaudiomodule.cpp b/contrib/python/pyrtaudio/rtaudiomodule.cpp
index 40a87e5961a38fb9a8353d6aca04438f45f931bc..93d3c73121c5ff98fcf5f999b276a98a42fc7bd1 100644 (file)
--- a/contrib/python/pyrtaudio/rtaudiomodule.cpp
+++ b/contrib/python/pyrtaudio/rtaudiomodule.cpp
@@ -48,7 +48,7 @@ extern "C" {
         PyObject *callback_func;
     } PyRtAudio;
 
-    static PyObject *RtAudioError;
+    static PyObject *RtAudioErrorException;
 
     static int callback(void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
         double streamTime, RtAudioStreamStatus status, void *data )
@@ -151,9 +151,9 @@ extern "C" {
             else if(!strcmp(api, "directsound"))
                 self->dac = new RtAudio(RtAudio::WINDOWS_DS);
         }
-        catch (RtError &error) {
-            PyErr_SetString(RtAudioError, error.getMessage().c_str());
-            Py_INCREF(RtAudioError);
+        catch (RtAudioError &error) {
+            PyErr_SetString(RtAudioErrorException, error.getMessage().c_str());
+            Py_INCREF(RtAudioErrorException);
             return NULL;
         }
 
@@ -261,9 +261,9 @@ extern "C" {
                 self->dac->closeStream();
             self->dac->openStream(&oParams, &iParams, self->_format, fs, &bf, &callback, self, &options);
         }
-        catch ( RtError& error ) {
-            PyErr_SetString(RtAudioError, error.getMessage().c_str());
-            Py_INCREF(RtAudioError);
+        catch ( RtAudioError& error ) {
+            PyErr_SetString(RtAudioErrorException, error.getMessage().c_str());
+            Py_INCREF(RtAudioErrorException);
             return NULL;
         }
 
@@ -281,9 +281,9 @@ extern "C" {
             self->dac->closeStream();
             Py_CLEAR(self->callback_func);
         }
-        catch(RtError &error) {
-            PyErr_SetString(RtAudioError, error.getMessage().c_str());
-            Py_INCREF(RtAudioError);
+        catch(RtAudioError &error) {
+            PyErr_SetString(RtAudioErrorException, error.getMessage().c_str());
+            Py_INCREF(RtAudioErrorException);
             return NULL;
         }
 
@@ -297,9 +297,9 @@ extern "C" {
         try {
             self->dac->startStream();
         }
-        catch(RtError &error) {
-            PyErr_SetString(RtAudioError, error.getMessage().c_str());
-            Py_INCREF(RtAudioError);
+        catch(RtAudioError &error) {
+            PyErr_SetString(RtAudioErrorException, error.getMessage().c_str());
+            Py_INCREF(RtAudioErrorException);
             return NULL;
         }
 
@@ -315,9 +315,9 @@ extern "C" {
         try {
             self->dac->stopStream();
         }
-        catch(RtError &error) {
-            PyErr_SetString(RtAudioError, error.getMessage().c_str());
-            Py_INCREF(RtAudioError);
+        catch(RtAudioError &error) {
+            PyErr_SetString(RtAudioErrorException, error.getMessage().c_str());
+            Py_INCREF(RtAudioErrorException);
             return NULL;
         }
 
@@ -332,9 +332,9 @@ extern "C" {
         try {
             self->dac->abortStream();
         }
-        catch(RtError &error) {
-            PyErr_SetString(RtAudioError, error.getMessage().c_str());
-            Py_INCREF(RtAudioError);
+        catch(RtAudioError &error) {
+            PyErr_SetString(RtAudioErrorException, error.getMessage().c_str());
+            Py_INCREF(RtAudioErrorException);
             return NULL;
         }
         Py_RETURN_NONE;
@@ -430,9 +430,9 @@ extern "C" {
             return info_dict;
 
         }
-        catch(RtError &error) {
-            PyErr_SetString(RtAudioError, error.getMessage().c_str());
-            Py_INCREF(RtAudioError);
+        catch(RtAudioError &error) {
+            PyErr_SetString(RtAudioErrorException, error.getMessage().c_str());
+            Py_INCREF(RtAudioErrorException);
             return NULL;
         }
     }
@@ -598,8 +598,8 @@ extern "C" {
         Py_INCREF(&RtAudio_type);
         PyModule_AddObject(module, "RtAudio", (PyObject *)&RtAudio_type);
 
-        RtAudioError = PyErr_NewException("rtaudio.RtError", NULL, NULL);
-        Py_INCREF(RtAudioError);
-        PyModule_AddObject(module, "RtError", RtAudioError);
+        RtAudioErrorException = PyErr_NewException("rtaudio.RtError", NULL, NULL);
+        Py_INCREF(RtAudioErrorException);
+        PyModule_AddObject(module, "RtError", RtAudioErrorException);
     }
 }

diff --git a/contrib/python/pyrtaudio/setup.py b/contrib/python/pyrtaudio/setup.py
index 6ad0d4a6889c5833dbc579b3f5f22b4fa002d414..a942f3b75aa2236e508115a5338e44f175329e4b 100644 (file)
--- a/contrib/python/pyrtaudio/setup.py
+++ b/contrib/python/pyrtaudio/setup.py
@@ -18,8 +18,7 @@ sources = ['rtaudiomodule.cpp', '../../../RtAudio.cpp']
 
 if OSNAME == 'Linux':
     define_macros=[("__LINUX_ALSA__", ''),
-                   ('__LINUX_JACK__', ''),
-                   ('__LINUX_OSS__', '')]
+                   ('__LINUX_JACK__', '')]
     libraries = ['asound', 'jack', 'pthread']
 
 elif OSNAME == 'Darwin':

diff --git a/doc/doxygen/compiling.txt b/doc/doxygen/compiling.txt
index 15f45b0a2b5efd9999d2acfb20729fac34d15685..29dcbd471afbf460e5d442521f0a818f166df6af 100644 (file)
--- a/doc/doxygen/compiling.txt
+++ b/doc/doxygen/compiling.txt
@@ -57,7 +57,7 @@ In order to compile RtAudio for a specific OS and audio API, it is necessary to
   <TD>RtApiCore</TD>
   <TD>__MACOSX_CORE__</TD>
   <TD><TT>pthread, CoreAudio</TT></TD>
-  <TD><TT>g++ -Wall -D__MACOSX_CORE__ -o audioprobe audioprobe.cpp RtAudio.cpp -framework CoreAudio -lpthread</TT></TD>
+  <TD><TT>g++ -Wall -D__MACOSX_CORE__ -o audioprobe audioprobe.cpp RtAudio.cpp -framework CoreAudio -framework CoreFoundation -lpthread</TT></TD>
 </TR>
 <TR>
   <TD>Windows</TD>

diff --git a/doc/doxygen/probe.txt b/doc/doxygen/probe.txt
index 653955b5498ed0dd121ab6f0324ae534f70bee21..975ff0eda069d7df9744c5d8eef5d63b3d1af97a 100644 (file)
--- a/doc/doxygen/probe.txt
+++ b/doc/doxygen/probe.txt
@@ -45,6 +45,7 @@ The RtAudio::DeviceInfo structure is defined in RtAudio.h and provides a variety
     bool isDefaultOutput;                  // true if this is the default output device.
     bool isDefaultInput;                   // true if this is the default input device.
     std::vector<unsigned int> sampleRates; // Supported sample rates.
+    unsigned int preferredSampleRate;      // Preferred sample rate, eg. for WASAPI the system sample rate.
     RtAudioFormat nativeFormats;           // Bit mask of supported data formats.
   };
 \endcode

diff --git a/tests/duplex.cpp b/tests/duplex.cpp
index 06462f2872aae5edbf6f883c01037cc8756f7df4..494e5ad7ab121e6ffa1ac98537d064af20c2c9f2 100644 (file)
--- a/tests/duplex.cpp
+++ b/tests/duplex.cpp
@@ -105,6 +105,7 @@ int main( int argc, char *argv[] )
   RtAudio::StreamOptions options;
   //options.flags |= RTAUDIO_NONINTERLEAVED;
 
+  bufferBytes = bufferFrames * channels * sizeof( MY_TYPE );
   try {
     adac.openStream( &oParams, &iParams, FORMAT, fs, &bufferFrames, &inout, (void *)&bufferBytes, &options );
   }
@@ -113,8 +114,6 @@ int main( int argc, char *argv[] )
     exit( 1 );
   }
 
-  bufferBytes = bufferFrames * channels * sizeof( MY_TYPE );
-
   // Test RtAudio functionality for reporting latency.
   std::cout << "\nStream latency = " << adac.getStreamLatency() << " frames" << std::endl;
 

diff --git a/tests/testall.cpp b/tests/testall.cpp
index 419f27b630729bfb9c6d5a3d99e6fdbc8832ccab..b94ce696d194114e1f36db55fe8a46e37134af13 100644 (file)
--- a/tests/testall.cpp
+++ b/tests/testall.cpp
@@ -66,7 +66,7 @@ int sawni( void *outputBuffer, void * /*inputBuffer*/, unsigned int nBufferFrame
   if ( status )
     std::cout << "Stream underflow detected!" << std::endl;
 
-  float increment;
+  double increment;
   for ( j=0; j<channels; j++ ) {
     increment = BASE_RATE * (j+1+(j*0.1));
     for ( i=0; i<nBufferFrames; i++ ) {

diff --git a/tests/teststops.cpp b/tests/teststops.cpp
index e24c06490ab0877b8ebfcf7500c00ba0404459a5..6159b88b00fb065c259cc58d948cb77b721d0a9f 100644 (file)
--- a/tests/teststops.cpp
+++ b/tests/teststops.cpp
@@ -63,7 +63,7 @@ int pulse( void *outputBuffer, void * /*inputBuffer*/, unsigned int nBufferFrame
   if ( status ) std::cout << "Stream over/underflow detected!" << std::endl;\r
 \r
   for ( i=0; i<nBufferFrames; i++ ) {\r
-    if ( data->frameCounter % data->pulseCount == 0 ) sample = 0.9;\r
+    if ( data->frameCounter % data->pulseCount == 0 ) sample = 0.9f;\r
     else sample = 0.0;\r
     for ( j=0; j<data->channels; j++ )\r
       *buffer++ = sample;\r
@@ -107,8 +107,8 @@ int main( int argc, char *argv[] )
   // Let RtAudio print messages to stderr.\r
   adc->showWarnings( true );\r
 \r
-  runtime = RUNTIME * 1000;\r
-  pausetime = PAUSETIME * 1000;\r
+  runtime = static_cast<unsigned int>(RUNTIME * 1000);\r
+  pausetime = static_cast<unsigned int>(PAUSETIME * 1000);\r
 \r
   // Set our stream parameters for a duplex stream.\r
   bufferFrames = 512;\r
@@ -127,7 +127,7 @@ int main( int argc, char *argv[] )
     oParams.deviceId = adc->getDefaultOutputDevice();\r
 \r
   // First, test external stopStream() calls.\r
-  mydata.pulseCount = PULSE_RATE * fs;\r
+  mydata.pulseCount = static_cast<unsigned int>(PULSE_RATE * fs);\r
   mydata.nFrames = 50 * fs;\r
   mydata.returnValue = 0;\r
   try {\r