Allow only the device's native sample rate in WASAPI API

Revert "fix" for Issue #69

RtAudio.cpp

@@ -3864,152 +3864,6 @@ private:
 
 //-----------------------------------------------------------------------------
 
-// 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 works best with conversions between one rate and its multiple.
-void convertBufferWasapi( char* outBuffer,
-                          const char* inBuffer,
-                          const unsigned int& channelCount,
-                          const unsigned int& inSampleRate,
-                          const unsigned int& outSampleRate,
-                          const unsigned int& inSampleCount,
-                          unsigned int& outSampleCount,
-                          const RtAudioFormat& format )
-{
-  // calculate the new outSampleCount and relative sampleStep
-  float sampleRatio = ( float ) outSampleRate / inSampleRate;
-  float sampleRatioInv = ( float ) 1 / sampleRatio;
-  float sampleStep = 1.0f / sampleRatio;
-  float inSampleFraction = 0.0f;
-
-  // for cmath functions
-  using namespace std;
-
-  outSampleCount = ( unsigned int ) roundf( inSampleCount * sampleRatio );
-
-  // if inSampleRate is a multiple of outSampleRate (or vice versa) there's no need to interpolate
-  if ( floor( sampleRatio ) == sampleRatio || floor( sampleRatioInv ) == sampleRatioInv )
-  {
-    // frame-by-frame, copy each relative input sample into it's corresponding output sample
-    for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
-    {
-      unsigned int inSample = ( unsigned int ) inSampleFraction;
-
-      switch ( format )
-      {
-        case RTAUDIO_SINT8:
-          memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );
-          break;
-        case RTAUDIO_SINT16:
-          memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );
-          break;
-        case RTAUDIO_SINT24:
-          memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );
-          break;
-        case RTAUDIO_SINT32:
-          memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );
-          break;
-        case RTAUDIO_FLOAT32:
-          memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );
-          break;
-        case RTAUDIO_FLOAT64:
-          memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
-          break;
-      }
-
-      // jump to next in sample
-      inSampleFraction += sampleStep;
-    }
-  }
-  else // else interpolate
-  {
-    // frame-by-frame, copy each relative input sample into it's corresponding output sample
-    for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
-    {
-      unsigned int inSample = ( unsigned int ) inSampleFraction;
-      float inSampleDec = inSampleFraction - inSample;
-      unsigned int frameInSample = inSample * channelCount;
-      unsigned int frameOutSample = outSample * channelCount;
-
-      switch ( format )
-      {
-        case RTAUDIO_SINT8:
-        {
-          for ( unsigned int channel = 0; channel < channelCount; channel++ )
-          {
-            char fromSample = ( ( char* ) inBuffer )[ frameInSample + channel ];
-            char toSample = ( ( char* ) inBuffer )[ frameInSample + channelCount + channel ];
-            char sampleDiff = ( char ) ( ( toSample - fromSample ) * inSampleDec );
-            ( ( char* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
-          }
-          break;
-        }
-        case RTAUDIO_SINT16:
-        {
-          for ( unsigned int channel = 0; channel < channelCount; channel++ )
-          {
-            short fromSample = ( ( short* ) inBuffer )[ frameInSample + channel ];
-            short toSample = ( ( short* ) inBuffer )[ frameInSample + channelCount + channel ];
-            short sampleDiff = ( short ) ( ( toSample - fromSample ) * inSampleDec );
-            ( ( short* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
-          }
-          break;
-        }
-        case RTAUDIO_SINT24:
-        {
-          for ( unsigned int channel = 0; channel < channelCount; channel++ )
-          {
-            int fromSample = ( ( S24* ) inBuffer )[ frameInSample + channel ].asInt();
-            int toSample = ( ( S24* ) inBuffer )[ frameInSample + channelCount + channel ].asInt();
-            int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );
-            ( ( S24* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
-          }
-          break;
-        }
-        case RTAUDIO_SINT32:
-        {
-          for ( unsigned int channel = 0; channel < channelCount; channel++ )
-          {
-            int fromSample = ( ( int* ) inBuffer )[ frameInSample + channel ];
-            int toSample = ( ( int* ) inBuffer )[ frameInSample + channelCount + channel ];
-            int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );
-            ( ( int* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
-          }
-          break;
-        }
-        case RTAUDIO_FLOAT32:
-        {
-          for ( unsigned int channel = 0; channel < channelCount; channel++ )
-          {
-            float fromSample = ( ( float* ) inBuffer )[ frameInSample + channel ];
-            float toSample = ( ( float* ) inBuffer )[ frameInSample + channelCount + channel ];
-            float sampleDiff = ( toSample - fromSample ) * inSampleDec;
-            ( ( float* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
-          }
-          break;
-        }
-        case RTAUDIO_FLOAT64:
-        {
-          for ( unsigned int channel = 0; channel < channelCount; channel++ )
-          {
-            double fromSample = ( ( double* ) inBuffer )[ frameInSample + channel ];
-            double toSample = ( ( double* ) inBuffer )[ frameInSample + channelCount + channel ];
-            double sampleDiff = ( toSample - fromSample ) * inSampleDec;
-            ( ( double* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
-          }
-          break;
-        }
-      }
-
-      // jump to next in sample
-      inSampleFraction += sampleStep;
-    }
-  }
-}
-
-//-----------------------------------------------------------------------------
-
 // A structure to hold various information related to the WASAPI implementation.
 struct WasapiHandle
 {
@@ -4275,15 +4129,12 @@ RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
     info.duplexChannels = 0;
   }
 
-  // sample rates
-  info.sampleRates.clear();
-
-  // 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++ ) {
-    info.sampleRates.push_back( SAMPLE_RATES[i] );
-  }
+  // sample rates (WASAPI only supports the one native sample rate)
   info.preferredSampleRate = deviceFormat->nSamplesPerSec;
 
+  info.sampleRates.clear();
+  info.sampleRates.push_back( deviceFormat->nSamplesPerSec );
+
   // native format
   info.nativeFormats = 0;
 
@@ -4559,6 +4410,7 @@ bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
   WAVEFORMATEX* deviceFormat = NULL;
   unsigned int bufferBytes;
   stream_.state = STREAM_STOPPED;
+  RtAudio::DeviceInfo deviceInfo;
 
   // create API Handle if not already created
   if ( !stream_.apiHandle )
@@ -4599,6 +4451,16 @@ bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
     goto Exit;
   }
 
+  deviceInfo = getDeviceInfo( device );
+
+  // validate sample rate
+  if ( sampleRate != deviceInfo.preferredSampleRate )
+  {
+    errorType = RtAudioError::INVALID_USE;
+    errorText_ = "RtApiWasapi::probeDeviceOpen: " + std::to_string( sampleRate ) + "Hz sample rate not supported. This device only supports " + std::to_string( deviceInfo.preferredSampleRate ) + "Hz.";
+    goto Exit;
+  }
+
   // determine whether index falls within capture or render devices
   if ( device >= renderDeviceCount ) {
     if ( mode != INPUT ) {
@@ -4682,7 +4544,7 @@ bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
   stream_.nUserChannels[mode] = channels;
   stream_.channelOffset[mode] = firstChannel;
   stream_.userFormat = format;
-  stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
+  stream_.deviceFormat[mode] = deviceInfo.nativeFormats;
 
   if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
     stream_.userInterleaved = false;
@@ -4782,8 +4644,6 @@ void RtApiWasapi::wasapiThread()
 
   WAVEFORMATEX* captureFormat = NULL;
   WAVEFORMATEX* renderFormat = NULL;
-  float captureSrRatio = 0.0f;
-  float renderSrRatio = 0.0f;
   WasapiBuffer captureBuffer;
   WasapiBuffer renderBuffer;
 
@@ -4793,15 +4653,11 @@ void RtApiWasapi::wasapiThread()
   unsigned long captureFlags = 0;
   unsigned int bufferFrameCount = 0;
   unsigned int numFramesPadding = 0;
-  unsigned int convBufferSize = 0;
   bool callbackPushed = false;
   bool callbackPulled = false;
   bool callbackStopped = false;
   int callbackResult = 0;
 
-  // convBuffer is used to store converted buffers between WASAPI and the user
-  char* convBuffer = NULL;
-  unsigned int convBuffSize = 0;
   unsigned int deviceBuffSize = 0;
 
   errorText_.clear();
@@ -4824,11 +4680,8 @@ void RtApiWasapi::wasapiThread()
       goto Exit;
     }
 
-    captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
-
     // initialize capture stream according to desire buffer size
-    float desiredBufferSize = stream_.bufferSize * captureSrRatio;
-    REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
+    REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) stream_.bufferSize * 10000000 / captureFormat->nSamplesPerSec );
 
     if ( !captureClient ) {
       hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
@@ -4875,7 +4728,7 @@ void RtApiWasapi::wasapiThread()
     }
 
     // scale outBufferSize according to stream->user sample rate ratio
-    unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
+    unsigned int outBufferSize = ( unsigned int ) stream_.bufferSize * stream_.nDeviceChannels[INPUT];
     inBufferSize *= stream_.nDeviceChannels[INPUT];
 
     // set captureBuffer size
@@ -4904,11 +4757,8 @@ void RtApiWasapi::wasapiThread()
       goto Exit;
     }
 
-    renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
-
     // initialize render stream according to desire buffer size
-    float desiredBufferSize = stream_.bufferSize * renderSrRatio;
-    REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
+    REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) stream_.bufferSize * 10000000 / renderFormat->nSamplesPerSec );
 
     if ( !renderClient ) {
       hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
@@ -4955,7 +4805,7 @@ void RtApiWasapi::wasapiThread()
     }
 
     // scale inBufferSize according to user->stream sample rate ratio
-    unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
+    unsigned int inBufferSize = ( unsigned int ) stream_.bufferSize * stream_.nDeviceChannels[OUTPUT];
     outBufferSize *= stream_.nDeviceChannels[OUTPUT];
 
     // set renderBuffer size
@@ -4978,23 +4828,18 @@ void RtApiWasapi::wasapiThread()
 
   if ( stream_.mode == INPUT ) {
     using namespace std; // for roundf
-    convBuffSize = ( size_t ) roundf( 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 ) {
-    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 ) {
-    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( convBuffSize );
   stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
-  if ( !convBuffer || !stream_.deviceBuffer ) {
+  if ( !stream_.deviceBuffer ) {
     errorType = RtAudioError::MEMORY_ERROR;
     errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
     goto Exit;
@@ -5006,26 +4851,15 @@ void RtApiWasapi::wasapiThread()
       // Callback Input
       // ==============
       // 1. Pull callback buffer from inputBuffer
-      // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
-      //                          Convert callback buffer to user format
+      // 2. If 1. was successful: Convert callback buffer to user format
 
       if ( captureAudioClient ) {
         // Pull callback buffer from inputBuffer
-        callbackPulled = captureBuffer.pullBuffer( convBuffer,
-                                                   ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],
+        callbackPulled = captureBuffer.pullBuffer( stream_.deviceBuffer,
+                                                   ( unsigned int ) stream_.bufferSize * stream_.nDeviceChannels[INPUT],
                                                    stream_.deviceFormat[INPUT] );
 
         if ( callbackPulled ) {
-          // Convert callback buffer to user sample rate
-          convertBufferWasapi( stream_.deviceBuffer,
-                               convBuffer,
-                               stream_.nDeviceChannels[INPUT],
-                               captureFormat->nSamplesPerSec,
-                               stream_.sampleRate,
-                               ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
-                               convBufferSize,
-                               stream_.deviceFormat[INPUT] );
-
           if ( stream_.doConvertBuffer[INPUT] ) {
             // Convert callback buffer to user format
             convertBuffer( stream_.userBuffer[INPUT],
@@ -5099,8 +4933,7 @@ void RtApiWasapi::wasapiThread()
     // Callback Output
     // ===============
     // 1. Convert callback buffer to stream format
-    // 2. Convert callback buffer to stream sample rate and channel count
-    // 3. Push callback buffer into outputBuffer
+    // 2. Push callback buffer into outputBuffer
 
     if ( renderAudioClient && callbackPulled ) {
       if ( stream_.doConvertBuffer[OUTPUT] ) {
@@ -5111,19 +4944,9 @@ void RtApiWasapi::wasapiThread()
 
       }
 
-      // 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],
+      callbackPushed = renderBuffer.pushBuffer( stream_.deviceBuffer,
+                                                stream_.bufferSize * stream_.nDeviceChannels[OUTPUT],
                                                 stream_.deviceFormat[OUTPUT] );
     }
     else {
@@ -5269,8 +5092,6 @@ Exit:
   CoTaskMemFree( captureFormat );
   CoTaskMemFree( renderFormat );
 
-  free ( convBuffer );
-
   CoUninitialize();
 
   // update stream state