diff --git a/examples/src/AudioDevice.cpp b/examples/src/AudioDevice.cpp
index ad1d91b..077e573 100644
--- a/examples/src/AudioDevice.cpp
+++ b/examples/src/AudioDevice.cpp
@@ -37,12 +37,12 @@ static int rt_callback(void * output_buffer, void * input_buffer, uint32_t num_b
 {
     if (status) std::cerr << "[rtaudio] buffer over or underflow" << std::endl;
 
-	// Playback
+    // Playback
     if (buffer.getAvailableRead()) buffer.read((float*) output_buffer, BUFFER_LENGTH);
     else memset(output_buffer, 0, BUFFER_LENGTH * sizeof(float));
 
-	// Recording
-	if (record_buffer.getAvailableWrite()) record_buffer.write((float*) input_buffer, BUFFER_LENGTH / 2);
+    // Recording
+    if (record_buffer.getAvailableWrite()) record_buffer.write((float*) input_buffer, BUFFER_LENGTH / 2);
 
     return 0;
 }
@@ -62,9 +62,9 @@ AudioDevice::~AudioDevice()
     {
         rtaudio->stopStream();
         if (rtaudio->isStreamOpen())
-		{
+        {
             rtaudio->closeStream();
-		}
+        }
     }
 }
 
@@ -77,7 +77,7 @@ bool AudioDevice::Open(const int deviceId)
     outputParams.nChannels = info.numChannels;
     outputParams.firstChannel = 0;
 
-	RtAudio::StreamParameters inputParams;
+    RtAudio::StreamParameters inputParams;
     inputParams.deviceId = rtaudio->getDefaultInputDevice();
     inputParams.nChannels = 1;
     inputParams.firstChannel = 0;
@@ -129,21 +129,21 @@ bool AudioDevice::Play(const std::vector<float> & data)
 
 bool AudioDevice::Record(const uint32_t lengthInSamples, std::vector<float> & recordingBuffer)
 {
-	uint64_t recordedSamples = 0;
+    uint64_t recordedSamples = 0;
 
-	// Allocate memory upfront (revisit this later...)
-	recordingBuffer.resize(lengthInSamples + (BUFFER_LENGTH)); // + a little padding
+    // Allocate memory upfront (revisit this later...)
+    recordingBuffer.resize(lengthInSamples + (BUFFER_LENGTH)); // + a little padding
 
-	while (recordedSamples < lengthInSamples)
-	{
-		if (record_buffer.getAvailableRead())
-		{
-			if (record_buffer.read(recordingBuffer.data() + recordedSamples, BUFFER_LENGTH / 2))
-			{
-				recordedSamples += (BUFFER_LENGTH / 2);
-			}
-		}
-	}
+    while (recordedSamples < lengthInSamples)
+    {
+        if (record_buffer.getAvailableRead())
+        {
+            if (record_buffer.read(recordingBuffer.data() + recordedSamples, BUFFER_LENGTH / 2))
+            {
+                recordedSamples += (BUFFER_LENGTH / 2);
+            }
+        }
+    }
 
-	return true;
+    return true;
 }
diff --git a/examples/src/AudioDevice.h b/examples/src/AudioDevice.h
index 02658be..618d6e5 100644
--- a/examples/src/AudioDevice.h
+++ b/examples/src/AudioDevice.h
@@ -39,27 +39,27 @@ static const int32_t BUFFER_LENGTH = FRAME_SIZE * CHANNELS;
 
 struct AudioDeviceInfo
 {
-	uint32_t id;
-	uint32_t numChannels;
-	uint32_t sampleRate;
-	uint32_t frameSize;
-	bool isPlaying = false;
+    uint32_t id;
+    uint32_t numChannels;
+    uint32_t sampleRate;
+    uint32_t frameSize;
+    bool isPlaying = false;
 };
 
 class AudioDevice
 {
-	std::unique_ptr<RtAudio> rtaudio;
+    std::unique_ptr<RtAudio> rtaudio;
 protected:
-	AudioDevice(const AudioDevice& r) = delete;
-	AudioDevice & operator = (const AudioDevice& r) = delete;
+    AudioDevice(const AudioDevice& r) = delete;
+    AudioDevice & operator = (const AudioDevice& r) = delete;
 public:
-	AudioDeviceInfo info;
-	AudioDevice(int numChannels, int sampleRate, int deviceId = -1);
-	~AudioDevice();
-	static void ListAudioDevices();
-	bool Open(const int deviceId);
-	bool Play(const std::vector<float> & data);
-	bool Record(const uint32_t lengthInSamples, std::vector<float> & recordingBuffer);
+    AudioDeviceInfo info;
+    AudioDevice(int numChannels, int sampleRate, int deviceId = -1);
+    ~AudioDevice();
+    static void ListAudioDevices();
+    bool Open(const int deviceId);
+    bool Play(const std::vector<float> & data);
+    bool Record(const uint32_t lengthInSamples, std::vector<float> & recordingBuffer);
 };
 
 #endif
diff --git a/examples/src/Main.cpp b/examples/src/Main.cpp
index 3d9faf2..0703c61 100644
--- a/examples/src/Main.cpp
+++ b/examples/src/Main.cpp
@@ -17,77 +17,77 @@ using namespace nqr;
 
 int main(int argc, const char **argv) try
 {
-	AudioDevice::ListAudioDevices();
+    AudioDevice::ListAudioDevices();
 
-	const int desiredSampleRate = 44100;
+    const int desiredSampleRate = 44100;
     const int desiredChannelCount = 2;
-	AudioDevice myDevice(desiredChannelCount, desiredSampleRate);
-	myDevice.Open(myDevice.info.id);
+    AudioDevice myDevice(desiredChannelCount, desiredSampleRate);
+    myDevice.Open(myDevice.info.id);
 
-	std::shared_ptr<AudioData> fileData = std::make_shared<AudioData>();
+    std::shared_ptr<AudioData> fileData = std::make_shared<AudioData>();
 
-	NyquistIO loader;
+    NyquistIO loader;
 
-	if (argc > 1)
-	{
-		std::string cli_arg = std::string(argv[1]);
-		loader.Load(fileData.get(), cli_arg);
-	}
-	else
-	{
-		// Circular libnyquist testing
-		//loader.Load(fileData.get(), "libnyquist_example_output.opus");
+    if (argc > 1)
+    {
+        std::string cli_arg = std::string(argv[1]);
+        loader.Load(fileData.get(), cli_arg);
+    }
+    else
+    {
+        // Circular libnyquist testing
+        //loader.Load(fileData.get(), "libnyquist_example_output.opus");
 
-		// 1-channel wave
-		//loader.Load(fileData.get(), "test_data/1ch/44100/8/test.wav");
-		//loader.Load(fileData.get(), "test_data/1ch/44100/16/test.wav");
-		//loader.Load(fileData.get(), "test_data/1ch/44100/24/test.wav");
-		//loader.Load(fileData.get(), "test_data/1ch/44100/32/test.wav");
-		//loader.Load(fileData.get(), "test_data/1ch/44100/64/test.wav");
+        // 1-channel wave
+        //loader.Load(fileData.get(), "test_data/1ch/44100/8/test.wav");
+        //loader.Load(fileData.get(), "test_data/1ch/44100/16/test.wav");
+        //loader.Load(fileData.get(), "test_data/1ch/44100/24/test.wav");
+        //loader.Load(fileData.get(), "test_data/1ch/44100/32/test.wav");
+        //loader.Load(fileData.get(), "test_data/1ch/44100/64/test.wav");
 
-		// 2-channel wave
-		//loader.Load(fileData.get(), "test_data/2ch/44100/8/test.wav");
-		//loader.Load(fileData.get(), "test_data/2ch/44100/16/test.wav");
-		//loader.Load(fileData.get(), "test_data/2ch/44100/24/test.wav");
-		//loader.Load(fileData.get(), "test_data/2ch/44100/32/test.wav");
-		//loader.Load(fileData.get(), "test_data/2ch/44100/64/test.wav");
+        // 2-channel wave
+        //loader.Load(fileData.get(), "test_data/2ch/44100/8/test.wav");
+        //loader.Load(fileData.get(), "test_data/2ch/44100/16/test.wav");
+        //loader.Load(fileData.get(), "test_data/2ch/44100/24/test.wav");
+        //loader.Load(fileData.get(), "test_data/2ch/44100/32/test.wav");
+        //loader.Load(fileData.get(), "test_data/2ch/44100/64/test.wav");
 
-		//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_44_16_mono-ima4-reaper.wav");
-		//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_44_16_stereo-ima4-reaper.wav");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_44_16_mono-ima4-reaper.wav");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_44_16_stereo-ima4-reaper.wav");
 
-		// Multi-channel wave
-		//loader.Load(fileData.get(), "test_data/ad_hoc/6_channel_44k_16b.wav");
+        // Multi-channel wave
+        //loader.Load(fileData.get(), "test_data/ad_hoc/6_channel_44k_16b.wav");
 
-		// 1 + 2 channel ogg
-		//loader.Load(fileData.get(), "test_data/ad_hoc/LR_Stereo.ogg");
-		//loader.Load(fileData.get(), "test_data/ad_hoc/TestLaugh_44k.ogg");
-		//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat.ogg");
-		//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeatMono.ogg");
-		//loader.Load(fileData.get(), "test_data/ad_hoc/BlockWoosh_Stereo.ogg");
+        // 1 + 2 channel ogg
+        //loader.Load(fileData.get(), "test_data/ad_hoc/LR_Stereo.ogg");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/TestLaugh_44k.ogg");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat.ogg");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/TestBeatMono.ogg");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/BlockWoosh_Stereo.ogg");
 
-		// 1 + 2 channel flac
-		//loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr8_Stereo_Dithered.flac");
-		//loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr16_Stereo.flac");
-		//loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr16_Mono.flac");
-		//loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr24_Stereo.flac");
+        // 1 + 2 channel flac
+        //loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr8_Stereo_Dithered.flac");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr16_Stereo.flac");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr16_Mono.flac");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/KittyPurr24_Stereo.flac");
 
         //auto memory = ReadFile("test_data/ad_hoc/KittyPurr24_Stereo.flac"); // broken
         //loader.Load(fileData.get(), "flac", memory.buffer); // broken
 
-		// Single-channel opus
-		//loader.Load(fileData.get(), "test_data/ad_hoc/detodos.opus"); // "Firefox: From All, To All"
+        // Single-channel opus
+        //loader.Load(fileData.get(), "test_data/ad_hoc/detodos.opus"); // "Firefox: From All, To All"
 
-		// 1 + 2 channel wavpack
-		//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Float32.wv");
-		//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Float32_Mono.wv");
-		//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int16.wv");
-		//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int24.wv");
-		//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int32.wv");
-		//loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int24_Mono.wv");
+        // 1 + 2 channel wavpack
+        //loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Float32.wv");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Float32_Mono.wv");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int16.wv");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int24.wv");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int32.wv");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/TestBeat_Int24_Mono.wv");
 
-		// 1 + 2 channel musepack
-		//loader.Load(fileData.get(), "test_data/ad_hoc/44_16_stereo.mpc");
-		//loader.Load(fileData.get(), "test_data/ad_hoc/44_16_mono.mpc");
+        // 1 + 2 channel musepack
+        //loader.Load(fileData.get(), "test_data/ad_hoc/44_16_stereo.mpc");
+        //loader.Load(fileData.get(), "test_data/ad_hoc/44_16_mono.mpc");
 
         // In-memory ogg
         //auto memory = ReadFile("test_data/ad_hoc/BlockWoosh_Stereo.ogg");
@@ -96,38 +96,38 @@ int main(int argc, const char **argv) try
         // In-memory Mp3
         auto memory = ReadFile("test_data/ad_hoc/acetylene.mp3");
         loader.Load(fileData.get(), "mp3", memory.buffer);
-	}
+    }
 
-	/* Test Recording Capabilities of AudioDevice
-	fileData->samples.reserve(44100 * 5);
-	fileData->channelCount = 1;
-	fileData->frameSize = 32;
-	fileData->lengthSeconds = 5.0;
-	fileData->sampleRate = 44100;
-	std::cout << "Starting recording ..." << std::endl;
-	myDevice.Record(fileData->sampleRate * fileData->lengthSeconds, fileData->samples);
-	*/
+    /* Test Recording Capabilities of AudioDevice
+    fileData->samples.reserve(44100 * 5);
+    fileData->channelCount = 1;
+    fileData->frameSize = 32;
+    fileData->lengthSeconds = 5.0;
+    fileData->sampleRate = 44100;
+    std::cout << "Starting recording ..." << std::endl;
+    myDevice.Record(fileData->sampleRate * fileData->lengthSeconds, fileData->samples);
+    */
 
-	if (fileData->sampleRate != desiredSampleRate)
-	{
-		std::cout << "[Warning - Sample Rate Mismatch] - file is sampled at " << fileData->sampleRate << " and output is " << desiredSampleRate << std::endl;
-	}
+    if (fileData->sampleRate != desiredSampleRate)
+    {
+        std::cout << "[Warning - Sample Rate Mismatch] - file is sampled at " << fileData->sampleRate << " and output is " << desiredSampleRate << std::endl;
+    }
 
-	std::cout << "Input Samples: " << fileData->samples.size() << std::endl;
+    std::cout << "Input Samples: " << fileData->samples.size() << std::endl;
 
-	// Convert mono to stereo for testing playback
-	if (fileData->channelCount == 1)
-	{
-		std::cout << "Playing MONO for: " << fileData->lengthSeconds << " seconds..." << std::endl;
-		std::vector<float> stereoCopy(fileData->samples.size() * 2);
-		MonoToStereo(fileData->samples.data(), stereoCopy.data(), fileData->samples.size());
-		myDevice.Play(stereoCopy);
-	}
-	else
-	{
-		std::cout << "Playing STEREO for: " << fileData->lengthSeconds << " seconds..." << std::endl;
-		myDevice.Play(fileData->samples);
-	}
+    // Convert mono to stereo for testing playback
+    if (fileData->channelCount == 1)
+    {
+        std::cout << "Playing MONO for: " << fileData->lengthSeconds << " seconds..." << std::endl;
+        std::vector<float> stereoCopy(fileData->samples.size() * 2);
+        MonoToStereo(fileData->samples.data(), stereoCopy.data(), fileData->samples.size());
+        myDevice.Play(stereoCopy);
+    }
+    else
+    {
+        std::cout << "Playing STEREO for: " << fileData->lengthSeconds << " seconds..." << std::endl;
+        myDevice.Play(fileData->samples);
+    }
 
     // Test Opus Encoding
     {
@@ -143,21 +143,21 @@ int main(int argc, const char **argv) try
         std::cout << "Encoder Status: " << encoderStatus << std::endl;
     }
 
-	return EXIT_SUCCESS;
+    return EXIT_SUCCESS;
 }
 catch (const UnsupportedExtensionEx & e)
 {
-	std::cerr << "Caught: " << e.what() << std::endl;
+    std::cerr << "Caught: " << e.what() << std::endl;
 }
 catch (const LoadPathNotImplEx & e)
 {
-	std::cerr << "Caught: " << e.what() << std::endl;
+    std::cerr << "Caught: " << e.what() << std::endl;
 }
 catch (const LoadBufferNotImplEx & e)
 {
-	std::cerr << "Caught: " << e.what() << std::endl;
+    std::cerr << "Caught: " << e.what() << std::endl;
 }
 catch (const std::exception & e)
 {
-	std::cerr << "Caught: " << e.what() << std::endl;
+    std::cerr << "Caught: " << e.what() << std::endl;
 }
\ No newline at end of file
diff --git a/examples/src/RingBuffer.h b/examples/src/RingBuffer.h
index dec2da5..8347a0e 100644
--- a/examples/src/RingBuffer.h
+++ b/examples/src/RingBuffer.h
@@ -45,7 +45,7 @@ class RingBufferT
   public:
 
     // Constructs a RingBufferT with size = 0
-	RingBufferT() : mData(nullptr), mAllocatedSize(0), mWriteIndex(0), mReadIndex(0) {}
+    RingBufferT() : mData(nullptr), mAllocatedSize(0), mWriteIndex(0), mReadIndex(0) {}
 
     // Constructs a RingBufferT with \a count maximum elements.
     RingBufferT(size_t count) : mAllocatedSize(0) { resize(count); }
diff --git a/include/libnyquist/Encoders.h b/include/libnyquist/Encoders.h
index b8a028d..e7024c0 100644
--- a/include/libnyquist/Encoders.h
+++ b/include/libnyquist/Encoders.h
@@ -1,44 +1,44 @@
 /*
-Copyright (c) 2019, Dimitri Diakopoulos All rights reserved.
+copyright (c) 2019, dimitri diakopoulos all rights reserved.
 
-Redistribution and use in source and binary forms, with or without
+redistribution and use in source and binary forms, with or without
 modification, are permitted provided that the following conditions are met:
 
-* Redistributions of source code must retain the above copyright notice, this
+* redistributions of source code must retain the above copyright notice, this
   list of conditions and the following disclaimer.
 
-* Redistributions in binary form must reproduce the above copyright notice,
+* redistributions in binary form must reproduce the above copyright notice,
   this list of conditions and the following disclaimer in the documentation
   and/or other materials provided with the distribution.
 
-THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
-AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
-IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
-DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
-FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
-DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
-SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
-CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
-OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+this software is provided by the copyright holders and contributors "as is"
+and any express or implied warranties, including, but not limited to, the
+implied warranties of merchantability and fitness for a particular purpose are
+disclaimed. in no event shall the copyright holder or contributors be liable
+for any direct, indirect, incidental, special, exemplary, or consequential
+damages (including, but not limited to, procurement of substitute goods or
+services; loss of use, data, or profits; or business interruption) however
+caused and on any theory of liability, whether in contract, strict liability,
+or tort (including negligence or otherwise) arising in any way out of the use
+of this software, even if advised of the possibility of such damage.
 */
 
-#ifndef NYQUIST_ENCODERS_H
-#define NYQUIST_ENCODERS_H
+#ifndef nyquist_encoders_h
+#define nyquist_encoders_h
 
-#include "Common.h"
+#include "common.h"
 
 namespace nqr
 {
-	// A simplistic encoder that takes a buffer of audio, conforms it to the user's
-	// EncoderParams preference, and writes to disk. Be warned, does not support resampling!
-	// @todo support dithering, samplerate conversion, etc.
-	int encode_wav_to_disk(const EncoderParams p, const AudioData * d, const std::string & path);
+    // a simplistic encoder that takes a buffer of audio, conforms it to the user's
+    // encoderparams preference, and writes to disk. be warned, does not support resampling!
+    // @todo support dithering, samplerate conversion, etc.
+    int encode_wav_to_disk(const encoderparams p, const audiodata * d, const std::string & path);
 
-    // Assume data adheres to EncoderParams, except for bit depth and fmt which are re-formatted
-    // to satisfy the Ogg/Opus spec.
-	int encode_opus_to_disk(const EncoderParams p, const AudioData * d, const std::string & path);
+    // assume data adheres to encoderparams, except for bit depth and fmt which are re-formatted
+    // to satisfy the ogg/opus spec.
+    int encode_opus_to_disk(const encoderparams p, const audiodata * d, const std::string & path);
 
 } // end namespace nqr
 
-#endif // end NYQUIST_ENCODERS_H
+#endif // end nyquist_encoders_h