#include <cstdlib>\r
#include <cstring>\r
#include <climits>\r
+#include <cmath>\r
#include <algorithm>\r
\r
// Static variable definitions.\r
//\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
throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );\r
}\r
\r
-RtAudio :: ~RtAudio() throw()\r
+RtAudio :: ~RtAudio()\r
{\r
if ( rtapi_ )\r
delete rtapi_;\r
\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
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
// 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
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
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
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
+ if ( floor( sampleRatio ) == sampleRatio || floor( sampleRatioInv ) == sampleRatioInv )\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
-\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
+ // 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
+\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
-\r
- switch (format)\r
- {\r
- case RTAUDIO_SINT8:\r
- {\r
- for (unsigned int channel = 0; channel < channelCount; channel++)\r
- {\r
- char fromSample = ((char*)inBuffer)[(inSample * channelCount) + channel];\r
- char toSample = ((char*)inBuffer)[((inSample + 1) * channelCount) + channel];\r
- float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor(inSampleFraction));\r
- ((char*)outBuffer)[(outSample * channelCount) + channel] = fromSample + (char)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)[(inSample * channelCount) + channel];\r
- short toSample = ((short*)inBuffer)[((inSample + 1) * channelCount) + channel];\r
- float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor(inSampleFraction));\r
- ((short*)outBuffer)[(outSample * channelCount) + channel] = fromSample + (short)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)[(inSample * channelCount) + channel].asInt();\r
- int toSample = ((S24*)inBuffer)[((inSample + 1) * channelCount) + channel].asInt();\r
- float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor(inSampleFraction));\r
- ((S24*)outBuffer)[(outSample * channelCount) + channel] = fromSample + (int)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)[(inSample * channelCount) + channel];\r
- int toSample = ((int*)inBuffer)[((inSample + 1) * channelCount) + channel];\r
- float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor(inSampleFraction));\r
- ((int*)outBuffer)[(outSample * channelCount) + channel] = fromSample + (int)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)[(inSample * channelCount) + channel];\r
- float toSample = ((float*)inBuffer)[((inSample + 1) * channelCount) + channel];\r
- float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor(inSampleFraction));\r
- ((float*)outBuffer)[(outSample * channelCount) + 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)[(inSample * channelCount) + channel];\r
- double toSample = ((double*)inBuffer)[((inSample + 1) * channelCount) + channel];\r
- double sampleDiff = (toSample - fromSample) * (inSampleFraction - floor(inSampleFraction));\r
- ((double*)outBuffer)[(outSample * channelCount) + channel] = fromSample + sampleDiff;\r
- }\r
- break;\r
- }\r
- }\r
-\r
- // jump to next in sample\r
- inSampleFraction += sampleStep;\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
+ }\r
}\r
}\r
\r
// 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
\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
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
}\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
projects / rtaudio-cdist.git / blobdiff