Merge pull request #68 from jpcima/master

[rtaudio-cdist.git] / RtAudio.cpp

diff --git a/RtAudio.cpp b/RtAudio.cpp
index f4bff00b0ae9d187d54f7e055d02ba466e88b48b..0b952f02ec8812c169d3eb17d1ca5c94ae151070 100644 (file)
--- a/RtAudio.cpp
+++ b/RtAudio.cpp
@@ -45,6 +45,7 @@
 #include <cstdlib>\r
 #include <cstring>\r
 #include <climits>\r
+#include <cmath>\r
 #include <algorithm>\r
 \r
 // Static variable definitions.\r
@@ -92,12 +93,12 @@ const unsigned int RtApi::SAMPLE_RATES[] = {
 //\r
 // *************************************************** //\r
 \r
-std::string RtAudio :: getVersion( void ) throw()\r
+std::string RtAudio :: getVersion( void )\r
 {\r
   return RTAUDIO_VERSION;\r
 }\r
 \r
-void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()\r
+void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )\r
 {\r
   apis.clear();\r
 \r
@@ -209,7 +210,7 @@ RtAudio :: RtAudio( RtAudio::Api api )
   throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );\r
 }\r
 \r
-RtAudio :: ~RtAudio() throw()\r
+RtAudio :: ~RtAudio()\r
 {\r
   if ( rtapi_ )\r
     delete rtapi_;\r
@@ -427,6 +428,9 @@ void RtApi :: setStreamTime( double time )
 \r
   if ( time >= 0.0 )\r
     stream_.streamTime = time;\r
+#if defined( HAVE_GETTIMEOFDAY )\r
+  gettimeofday( &stream_.lastTickTimestamp, NULL );\r
+#endif\r
 }\r
 \r
 unsigned int RtApi :: getStreamSampleRate( void )\r
@@ -1943,7 +1947,7 @@ struct JackHandle {
 static void jackSilentError( const char * ) {};\r
 \r
 RtApiJack :: RtApiJack()\r
-{\r
+    :shouldAutoconnect_(true) {\r
   // Nothing to do here.\r
 #if !defined(__RTAUDIO_DEBUG__)\r
   // Turn off Jack's internal error reporting.\r
@@ -2354,6 +2358,8 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
   // here.\r
   if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );\r
 \r
+  if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false;\r
+\r
   return SUCCESS;\r
 \r
  error:\r
@@ -2443,7 +2449,7 @@ void RtApiJack :: startStream( void )
   const char **ports;\r
 \r
   // Get the list of available ports.\r
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
+  if ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) {\r
     result = 1;\r
     ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);\r
     if ( ports == NULL) {\r
@@ -2467,7 +2473,7 @@ void RtApiJack :: startStream( void )
     free(ports);\r
   }\r
 \r
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
+  if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) {\r
     result = 1;\r
     ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );\r
     if ( ports == NULL) {\r
@@ -3875,7 +3881,7 @@ void convertBufferWasapi( char* outBuffer,
   float sampleStep = 1.0f / sampleRatio;\r
   float inSampleFraction = 0.0f;\r
 \r
-  outSampleCount = ( unsigned int ) roundf( inSampleCount * sampleRatio );\r
+  outSampleCount = ( unsigned int ) std::roundf( inSampleCount * sampleRatio );\r
 \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
@@ -5283,6 +5289,8 @@ Exit:
 // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007\r
 // Changed device query structure for RtAudio 4.0.7, January 2010\r
 \r
+#include <mmsystem.h>\r
+#include <mmreg.h>\r
 #include <dsound.h>\r
 #include <assert.h>\r
 #include <algorithm>\r
@@ -5363,8 +5371,8 @@ RtApiDs :: RtApiDs()
 \r
 RtApiDs :: ~RtApiDs()\r
 {\r
-  if ( coInitialized_ ) CoUninitialize(); // balanced call.\r
   if ( stream_.state != STREAM_CLOSED ) closeStream();\r
+  if ( coInitialized_ ) CoUninitialize(); // balanced call.\r
 }\r
 \r
 // The DirectSound default output is always the first device.\r
@@ -8783,8 +8791,10 @@ 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
@@ -9111,7 +9121,7 @@ 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