diff --git a/src/FlacDecoder.cpp b/src/FlacDecoder.cpp
index a0970a0..f2b6ea6 100644
--- a/src/FlacDecoder.cpp
+++ b/src/FlacDecoder.cpp
@@ -75,26 +75,6 @@ public:
             d->lengthSeconds = (float) numSamples / (float) d->sampleRate;
             
             auto totalSamples = numSamples * d->channelCount;
-            
-			// N.B.: "Currently the reference encoder and decoders only support up to 24 bits per sample."
-
-            d->sourceFormat = PCMFormat::PCM_END;
-
-			switch (d->bitDepth)
-			{
-			case 8:
-				internalFmt = PCMFormat::PCM_S8;
-				break;
-			case 16:
-				internalFmt = PCMFormat::PCM_16;
-				break;
-			case 24:
-				internalFmt = PCMFormat::PCM_24;
-				break;
-			default:
-				throw std::runtime_error("unsupported FLAC bit depth");
-				break;
-			}
 
             // Next, process internal buffer into the user-visible samples array
             ConvertToFloat32(d->samples.data(), internalBuffer.data(), totalSamples, d->sourceFormat);
@@ -118,12 +98,13 @@ public:
     
     void processMetadata(const FLAC__StreamMetadata_StreamInfo & info)
     {
+        // N.B.: "Currently the reference encoder and decoders only support up to 24 bits per sample."
         d->sampleRate = info.sample_rate;
         d->channelCount = info.channels; // Assert 1 to 8
-        d->sourceF = info.bits_per_sample; // Assert 4 to 32
+        d->sourceFormat = MakeFormatForBits(info.bits_per_sample, false, true);
         d->frameSize = info.channels * info.bits_per_sample;
         
-        const size_t bytesPerSample = d->bitDepth / 8;
+        const size_t bytesPerSample = GetFormatBitsPerSample(d->sourceFormat) / 8;
 
         numSamples = (size_t) info.total_samples;
         
@@ -139,7 +120,7 @@ public:
     {
         FlacDecoderInternal * decoder = reinterpret_cast<FlacDecoderInternal *>(userPtr);
         
-        const size_t bytesPerSample = decoder->d->bitDepth / 8;
+        const size_t bytesPerSample = GetFormatBitsPerSample(decoder->d->sourceFormat) / 8;
         
         auto dataPtr = decoder->internalBuffer.data();
         
diff --git a/src/OpusDecoder.cpp b/src/OpusDecoder.cpp
index 0335222..acd5fe5 100644
--- a/src/OpusDecoder.cpp
+++ b/src/OpusDecoder.cpp
@@ -69,9 +69,9 @@ public:
         
         d->sampleRate = OPUS_SAMPLE_RATE;
         d->channelCount = (uint32_t) header->channel_count;
-        d->bitDepth = 32;
+        d->sourceFormat = MakeFormatForBits(32, true, false);
         d->lengthSeconds = double(getLengthInSeconds());
-        d->frameSize = (uint32_t) header->channel_count * d->bitDepth;
+        d->frameSize = (uint32_t) header->channel_count * GetFormatBitsPerSample(d->sourceFormat);
         
         // Samples in a single channel
         auto totalSamples = size_t(getTotalSamples());
diff --git a/src/VorbisDecoder.cpp b/src/VorbisDecoder.cpp
index fda0454..aab927b 100644
--- a/src/VorbisDecoder.cpp
+++ b/src/VorbisDecoder.cpp
@@ -80,9 +80,9 @@ public:
         
         d->sampleRate = int(ovInfo->rate);
         d->channelCount = ovInfo->channels;
-        d->bitDepth = 32; // float
+        d->sourceFormat = MakeFormatForBits(32, true, false);
         d->lengthSeconds = double(getLengthInSeconds());
-        d->frameSize = ovInfo->channels * d->bitDepth;
+        d->frameSize = ovInfo->channels * GetFormatBitsPerSample(d->sourceFormat);
         
         // Samples in a single channel
         auto totalSamples = size_t(getTotalSamples());
diff --git a/src/WavDecoder.cpp b/src/WavDecoder.cpp
index 133be4b..e1c8c2b 100644
--- a/src/WavDecoder.cpp
+++ b/src/WavDecoder.cpp
@@ -178,28 +178,26 @@ int WavDecoder::LoadFromBuffer(AudioData * data, const std::vector<uint8_t> & me
     size_t totalSamples = (DataChunkInfo.size / wavHeader.frame_size) * wavHeader.channel_count;
     data->samples.resize(totalSamples);
 
-    PCMFormat internalFmt = PCMFormat::PCM_END;
-
 	switch (bit_depth)
 	{
 	case 8:
-		internalFmt = PCMFormat::PCM_U8;
+		data->sourceFormat = PCMFormat::PCM_U8;
 		break;
 	case 16:
-		internalFmt = PCMFormat::PCM_16;
+		data->sourceFormat = PCMFormat::PCM_16;
 		break;
 	case 24:
-		internalFmt = PCMFormat::PCM_24;
+		data->sourceFormat = PCMFormat::PCM_24;
 		break;
 	case 32:
-		internalFmt = (wavHeader.format == WaveFormatCode::FORMAT_IEEE) ? PCMFormat::PCM_FLT : PCMFormat::PCM_32;
+		data->sourceFormat = (wavHeader.format == WaveFormatCode::FORMAT_IEEE) ? PCMFormat::PCM_FLT : PCMFormat::PCM_32;
 		break;
 	case 64:
-		internalFmt = (wavHeader.format == WaveFormatCode::FORMAT_IEEE) ? PCMFormat::PCM_DBL : PCMFormat::PCM_64;
+		data->sourceFormat = (wavHeader.format == WaveFormatCode::FORMAT_IEEE) ? PCMFormat::PCM_DBL : PCMFormat::PCM_64;
 		break;
 	}
 
-	ConvertToFloat32(data->samples.data(), memory.data() + DataChunkInfo.offset, totalSamples, internalFmt);
+	ConvertToFloat32(data->samples.data(), memory.data() + DataChunkInfo.offset, totalSamples, data->sourceFormat);
     
     return IOError::NoError;
 }
diff --git a/src/WavPackDecoder.cpp b/src/WavPackDecoder.cpp
index bd40e52..63d4160 100644
--- a/src/WavPackDecoder.cpp
+++ b/src/WavPackDecoder.cpp
@@ -43,38 +43,23 @@ public:
             throw std::runtime_error("Not a WavPack file");
         }
         
+        auto bitdepth = WavpackGetBitsPerSample(context);
+        
         d->sampleRate = WavpackGetSampleRate(context);
         d->channelCount = WavpackGetNumChannels(context);
-        d->bitDepth = WavpackGetBitsPerSample(context);
         d->lengthSeconds = double(getLengthInSeconds());
-        d->frameSize = d->channelCount * d->bitDepth;
+        d->frameSize = d->channelCount * bitdepth;
         
         //@todo support channel masks
         // WavpackGetChannelMask
         
         auto totalSamples = size_t(getTotalSamples());
         
-        PCMFormat internalFmt = PCMFormat::PCM_END;
-        
         int mode = WavpackGetMode(context);
         int isFloatingPoint = (MODE_FLOAT & mode);
         
-        switch (d->bitDepth)
-        {
-            case 16:
-                internalFmt = PCMFormat::PCM_16;
-                break;
-            case 24:
-                internalFmt = PCMFormat::PCM_24;
-                break;
-            case 32:
-                internalFmt = isFloatingPoint ? PCMFormat::PCM_FLT : PCMFormat::PCM_32;
-                break;
-            default:
-                throw std::runtime_error("unsupported WavPack bit depth");
-                break;
-        }
-        
+        d->sourceFormat = MakeFormatForBits(bitdepth, isFloatingPoint, false);
+
         /* From the docs:
             "... required memory at "buffer" is 4 * samples * num_channels bytes. The
             audio data is returned right-justified in 32-bit longs in the endian
@@ -89,7 +74,7 @@ public:
         
         // Next, process internal buffer into the user-visible samples array
         if (!isFloatingPoint)
-            ConvertToFloat32(d->samples.data(), internalBuffer.data(), totalSamples * d->channelCount, internalFmt);
+            ConvertToFloat32(d->samples.data(), internalBuffer.data(), totalSamples * d->channelCount, d->sourceFormat);
         
     }