Merge branch 'master' of https://github.com/thestk/rtaudio

RtAudio.cpp

@@ -3585,7 +3585,7 @@ static const char* getAsioErrorString( ASIOError result )
 // - 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
+// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
 
 #ifndef INITGUID
   #define INITGUID
@@ -3767,15 +3767,14 @@ private:
 
 //-----------------------------------------------------------------------------
 
-// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate and
-// 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.
+// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
+// between HW and the user. The convertBufferWasapi function is used to perform this conversion
+// 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.
 void convertBufferWasapi( char* outBuffer,
                           const char* inBuffer,
-                          const unsigned int& inChannelCount,
-                          const unsigned int& outChannelCount,
+                          const unsigned int& channelCount,
                           const unsigned int& inSampleRate,
                           const unsigned int& outSampleRate,
                           const unsigned int& inSampleCount,
@@ -3786,7 +3785,6 @@ void convertBufferWasapi( char* outBuffer,
   float sampleRatio = ( float ) outSampleRate / inSampleRate;
   float sampleStep = 1.0f / sampleRatio;
   float inSampleFraction = 0.0f;
-  unsigned int commonChannelCount = std::min( inChannelCount, outChannelCount );
 
   outSampleCount = ( unsigned int ) ( inSampleCount * sampleRatio );
 
@@ -3798,22 +3796,22 @@ void convertBufferWasapi( char* outBuffer,
     switch ( format )
     {
       case RTAUDIO_SINT8:
-        memcpy( &( ( char* ) outBuffer )[ outSample * outChannelCount ], &( ( char* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( char ) );
+        memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );
         break;
       case RTAUDIO_SINT16:
-        memcpy( &( ( short* ) outBuffer )[ outSample * outChannelCount ], &( ( short* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( short ) );
+        memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );
         break;
       case RTAUDIO_SINT24:
-        memcpy( &( ( S24* ) outBuffer )[ outSample * outChannelCount ], &( ( S24* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( S24 ) );
+        memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );
         break;
       case RTAUDIO_SINT32:
-        memcpy( &( ( int* ) outBuffer )[ outSample * outChannelCount ], &( ( int* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( int ) );
+        memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );
         break;
       case RTAUDIO_FLOAT32:
-        memcpy( &( ( float* ) outBuffer )[ outSample * outChannelCount ], &( ( float* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( float ) );
+        memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );
         break;
       case RTAUDIO_FLOAT64:
-        memcpy( &( ( double* ) outBuffer )[ outSample * outChannelCount ], &( ( double* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( double ) );
+        memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
         break;
     }
 
@@ -4508,10 +4506,11 @@ bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
 
   // Set flags for buffer conversion.
   stream_.doConvertBuffer[mode] = false;
-  if ( stream_.userFormat != stream_.deviceFormat[mode] )
+  if ( stream_.userFormat != stream_.deviceFormat[mode] ||
+       stream_.nUserChannels != stream_.nDeviceChannels )
     stream_.doConvertBuffer[mode] = true;
-  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-       stream_.nUserChannels[mode] > 1 )
+  else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+            stream_.nUserChannels[mode] > 1 )
     stream_.doConvertBuffer[mode] = true;
 
   if ( stream_.doConvertBuffer[mode] )
@@ -4830,11 +4829,10 @@ void RtApiWasapi::wasapiThread()
                                                    stream_.deviceFormat[INPUT] );
 
         if ( callbackPulled ) {
-          // Convert callback buffer to user sample rate and channel count
+          // Convert callback buffer to user sample rate
           convertBufferWasapi( stream_.deviceBuffer,
                                convBuffer,
                                stream_.nDeviceChannels[INPUT],
-                               stream_.nUserChannels[INPUT],
                                captureFormat->nSamplesPerSec,
                                stream_.sampleRate,
                                ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
@@ -4848,7 +4846,7 @@ void RtApiWasapi::wasapiThread()
                            stream_.convertInfo[INPUT] );
           }
           else {
-            // no conversion, simple copy deviceBuffer to userBuffer
+            // no further conversion, simple copy deviceBuffer to userBuffer
             memcpy( stream_.userBuffer[INPUT],
                     stream_.deviceBuffer,
                     stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
@@ -4924,35 +4922,27 @@ void RtApiWasapi::wasapiThread()
                        stream_.userBuffer[OUTPUT],
                        stream_.convertInfo[OUTPUT] );
 
-        // Convert callback buffer to stream sample rate and channel count
-        convertBufferWasapi( convBuffer,
-                             stream_.deviceBuffer,
-                             stream_.nUserChannels[OUTPUT],
-                             stream_.nDeviceChannels[OUTPUT],
-                             stream_.sampleRate,
-                             renderFormat->nSamplesPerSec,
-                             stream_.bufferSize,
-                             convBufferSize,
-                             stream_.deviceFormat[OUTPUT] );
-      }
-      else {
-        // Convert callback buffer to stream sample rate and channel count
-        convertBufferWasapi( convBuffer,
-                             stream_.userBuffer[OUTPUT],
-                             stream_.nUserChannels[OUTPUT],
-                             stream_.nDeviceChannels[OUTPUT],
-                             stream_.sampleRate,
-                             renderFormat->nSamplesPerSec,
-                             stream_.bufferSize,
-                             convBufferSize,
-                             stream_.deviceFormat[OUTPUT] );
       }
 
+      // Convert callback buffer to stream sample rate
+      convertBufferWasapi( convBuffer,
+                           stream_.deviceBuffer,
+                           stream_.nDeviceChannels[OUTPUT],
+                           stream_.sampleRate,
+                           renderFormat->nSamplesPerSec,
+                           stream_.bufferSize,
+                           convBufferSize,
+                           stream_.deviceFormat[OUTPUT] );
+
       // Push callback buffer into outputBuffer
       callbackPushed = renderBuffer.pushBuffer( convBuffer,
                                                 convBufferSize * stream_.nDeviceChannels[OUTPUT],
                                                 stream_.deviceFormat[OUTPUT] );
     }
+    else {
+      // if there is no render stream, set callbackPushed flag
+      callbackPushed = true;
+    }
 
     // Stream Capture
     // ==============
@@ -4978,8 +4968,8 @@ void RtApiWasapi::wasapiThread()
       if ( bufferFrameCount != 0 ) {
         // Push capture buffer into inputBuffer
         if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
-                                      bufferFrameCount * stream_.nDeviceChannels[INPUT],
-                                      stream_.deviceFormat[INPUT] ) )
+                                       bufferFrameCount * stream_.nDeviceChannels[INPUT],
+                                       stream_.deviceFormat[INPUT] ) )
         {
           // Release capture buffer
           hr = captureClient->ReleaseBuffer( bufferFrameCount );
@@ -5047,8 +5037,8 @@ void RtApiWasapi::wasapiThread()
         // Pull next buffer from outputBuffer
         // Fill render buffer with next buffer
         if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
-                                     bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
-                                     stream_.deviceFormat[OUTPUT] ) )
+                                      bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
+                                      stream_.deviceFormat[OUTPUT] ) )
         {
           // Release render buffer
           hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );