-/************************************************************************/\r
+/************************************************************************/\r
/*! \class RtAudio\r
\brief Realtime audio i/o C++ classes.\r
\r
RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/\r
\r
RtAudio: realtime audio i/o C++ classes\r
- Copyright (c) 2001-2014 Gary P. Scavone\r
+ Copyright (c) 2001-2016 Gary P. Scavone\r
\r
Permission is hereby granted, free of charge, to any person\r
obtaining a copy of this software and associated documentation files\r
*/\r
/************************************************************************/\r
\r
-// RtAudio: Version 4.1.1\r
+// RtAudio: Version 4.1.2\r
\r
#include "RtAudio.h"\r
#include <iostream>\r
}\r
\r
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
if (handle) {\r
AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,\r
// 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
-void convertBufferWasapi( char* outBuffer,\r
- const char* inBuffer,\r
- const unsigned int& channelCount,\r
- const unsigned int& inSampleRate,\r
- const unsigned int& outSampleRate,\r
- const unsigned int& inSampleCount,\r
- unsigned int& outSampleCount,\r
- const RtAudioFormat& format )\r
-{\r
- // calculate the new outSampleCount and relative sampleStep\r
- float sampleRatio = ( float ) outSampleRate / inSampleRate;\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
- {\r
- unsigned int inSample = ( unsigned int ) inSampleFraction;\r
-\r
- switch ( format )\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
+ const unsigned int& inSampleRate,\r
+ const unsigned int& outSampleRate,\r
+ const unsigned int& inSampleCount,\r
+ unsigned int& outSampleCount,\r
+ const RtAudioFormat& format)\r
+{\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
+ // 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
- 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
+ // 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
+ // jump to next in sample\r
+ inSampleFraction += sampleStep;\r
+ }\r
+ }\r
}\r
\r
//-----------------------------------------------------------------------------\r
// if the callback buffer was pushed renderBuffer reset callbackPulled flag\r
if ( callbackPushed ) {\r
callbackPulled = false;\r
+ // tick stream time\r
+ RtApi::tickStreamTime();\r
}\r
\r
- // tick stream time\r
- RtApi::tickStreamTime();\r
}\r
\r
Exit:\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";\r
errorText_ = errorStream_.str();\r
+ MUTEX_UNLOCK( &stream_.mutex );\r
error( RtAudioError::SYSTEM_ERROR );\r
return;\r
}\r
bool *isRunning = &info->isRunning;\r
\r
#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)\r
- if ( &info->doRealtime ) {\r
+ if ( info->doRealtime ) {\r
pthread_t tID = pthread_self(); // ID of this thread\r
sched_param prio = { info->priority }; // scheduling priority of thread\r
pthread_setschedparam( tID, SCHED_RR, &prio );\r
}\r
break;\r
case OUTPUT:\r
- pah->s_play = pa_simple_new( NULL, "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );\r
+ pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );\r
if ( !pah->s_play ) {\r
errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";\r
goto error;\r
\r
void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )\r
{\r
- register char val;\r
- register char *ptr;\r
+ char val;\r
+ char *ptr;\r
\r
ptr = buffer;\r
if ( format == RTAUDIO_SINT16 ) {\r
projects / rtaudio-cdist.git / blobdiff