Merge pull request #8 from MarcusTomlinson/master

Fix WASAPI shutdown crash

1 files changed, 21 insertions, 20 deletions

diff --git a/RtAudio.cpp b/RtAudio.cpp
index dd06b52..9760c7c 100644
--- a/RtAudio.cpp
+++ b/RtAudio.cpp
@@ -3578,6 +3578,12 @@ static const char* getAsioErrorString( ASIOError result )
 

 #if defined(__WINDOWS_WASAPI__) // Windows WASAPI API

 

+// Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014

+// - Introduces support for the Windows WASAPI API

+// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required

+// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface

+// - Includes automatic internal conversion of sample rate, buffer size and channel count

+

 #ifndef INITGUID

   #define INITGUID

 #endif

@@ -3762,8 +3768,7 @@ private:
 // channel counts between HW and the user. The convertBufferWasapi function is used to perform

 // these conversions between HwIn->UserIn and UserOut->HwOut during the stream callback loop.

 // This sample rate converter favors speed over quality, and works best with conversions between

-// one rate and its multiple. RtApiWasapi will not populate a device's sample rate list with rates

-// that may cause artifacts via this conversion.

+// one rate and its multiple.

 void convertBufferWasapi( char* outBuffer,

                           const char* inBuffer,

                           const unsigned int& inChannelCount,

@@ -4090,18 +4095,9 @@ RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
   // sample rates

   info.sampleRates.clear();

 

-  // allow support for sample rates that are multiples of the base rate

+  // allow support for all sample rates as we have a built-in sample rate converter

   for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {

-    if ( SAMPLE_RATES[i] < deviceFormat->nSamplesPerSec ) {

-      if ( deviceFormat->nSamplesPerSec % SAMPLE_RATES[i] == 0 ) {

-        info.sampleRates.push_back( SAMPLE_RATES[i] );

-      }

-    }

-    else {

-      if ( SAMPLE_RATES[i] % deviceFormat->nSamplesPerSec == 0 ) {

-        info.sampleRates.push_back( SAMPLE_RATES[i] );

-      }

-    }

+    info.sampleRates.push_back( SAMPLE_RATES[i] );

   }

 

   // native format

@@ -4620,7 +4616,8 @@ void RtApiWasapi::wasapiThread()
 

   // convBuffer is used to store converted buffers between WASAPI and the user

   char* convBuffer = NULL;

-  unsigned int deviceBufferSize = 0;

+  unsigned int convBuffSize = 0;

+  unsigned int deviceBuffSize = 0;

 

   errorText_.clear();

   RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;

@@ -4795,18 +4792,22 @@ void RtApiWasapi::wasapiThread()
   }

 

   if ( stream_.mode == INPUT ) {

-    deviceBufferSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );

+    convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );

+    deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );

   }

   else if ( stream_.mode == OUTPUT ) {

-    deviceBufferSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );

+    convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );

+    deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );

   }

   else if ( stream_.mode == DUPLEX ) {

-    deviceBufferSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),

-                            ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );

+    convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),

+                             ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );

+    deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),

+                               stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );

   }

 

-  convBuffer = ( char* ) malloc( deviceBufferSize );

-  stream_.deviceBuffer = ( char* ) malloc( deviceBufferSize );

+  convBuffer = ( char* ) malloc( convBuffSize );

+  stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );

   if ( !convBuffer || !stream_.deviceBuffer ) {

     errorType = RtAudioError::MEMORY_ERROR;

     errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";