diff --git a/include/libnyquist/Common.h b/include/libnyquist/Common.h
index 7e9101d..54ec118 100644
--- a/include/libnyquist/Common.h
+++ b/include/libnyquist/Common.h
@@ -221,10 +221,11 @@ struct AudioData
 {
 	int channelCount;
 	int sampleRate;
-    int bitDepth;
 	double lengthSeconds;
 	size_t frameSize; // channels * bits per sample
 	std::vector<float> samples;
+    PCMFormat sourceFormat;
+    
     //@todo: add field: channel layout
     //@todo: add field: lossy / lossless
     //@todo: audio data loaded (for metadata only)
@@ -247,6 +248,7 @@ struct NyquistFileBuffer
 NyquistFileBuffer ReadFile(std::string pathToFile);
     
 int GetFormatBitsPerSample(PCMFormat f);
+PCMFormat MakeFormatForBits(int bits, bool floatingPt, bool isSigned);
 
 } // end namespace nqr
 
diff --git a/src/Common.cpp b/src/Common.cpp
index 33c5a60..4b6742c 100644
--- a/src/Common.cpp
+++ b/src/Common.cpp
@@ -167,3 +167,22 @@ int nqr::GetFormatBitsPerSample(PCMFormat f)
             return 0;
     }
 }
+
+PCMFormat nqr::MakeFormatForBits(int bits, bool floatingPt, bool isSigned)
+{
+    switch(bits)
+    {
+        case 8:
+            return (isSigned) ? PCM_S8 : PCM_U8;
+        case 16:
+            return PCM_16;
+        case 24:
+            return PCM_24;
+        case 32:
+            return (floatingPt) ? PCM_FLT : PCM_32;
+        case 64:
+            return (floatingPt) ? PCM_DBL : PCM_64;
+        default:
+            return PCM_END;
+    }
+}
diff --git a/src/FlacDecoder.cpp b/src/FlacDecoder.cpp
index 7809220..a0970a0 100644
--- a/src/FlacDecoder.cpp
+++ b/src/FlacDecoder.cpp
@@ -78,7 +78,7 @@ public:
             
 			// N.B.: "Currently the reference encoder and decoders only support up to 24 bits per sample."
 
-            PCMFormat internalFmt = PCMFormat::PCM_END;
+            d->sourceFormat = PCMFormat::PCM_END;
 
 			switch (d->bitDepth)
 			{
@@ -97,7 +97,7 @@ public:
 			}
 
             // Next, process internal buffer into the user-visible samples array
-            ConvertToFloat32(d->samples.data(), internalBuffer.data(), totalSamples, internalFmt);
+            ConvertToFloat32(d->samples.data(), internalBuffer.data(), totalSamples, d->sourceFormat);
         }
         
         else
@@ -120,7 +120,7 @@ public:
     {
         d->sampleRate = info.sample_rate;
         d->channelCount = info.channels; // Assert 1 to 8
-        d->bitDepth = info.bits_per_sample; // Assert 4 to 32
+        d->sourceF = info.bits_per_sample; // Assert 4 to 32
         d->frameSize = info.channels * info.bits_per_sample;
         
         const size_t bytesPerSample = d->bitDepth / 8;
diff --git a/src/WavEncoder.cpp b/src/WavEncoder.cpp
index 03a54aa..2784912 100644
--- a/src/WavEncoder.cpp
+++ b/src/WavEncoder.cpp
@@ -106,7 +106,7 @@ int WavEncoder::WriteFile(const EncoderParams p, const AudioData * d, const std:
     
     // Debugging -- assume IEEE_Float
     
-    auto LookupB
+    auto what = GetFormatBitsPerSample(d->sourceFmt);
     fout.write(reinterpret_cast<const char*>(d->samples.data()), numSamplesBytes);
     
     // Find size