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

1 files changed, 35 insertions, 45 deletions

diff --git a/RtAudio.cpp b/RtAudio.cpp
index a31a54b..9f12016 100644
--- a/RtAudio.cpp
+++ b/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 );