From 82939a2228fb00c243423bc72ffd31ca37e81074 Mon Sep 17 00:00:00 2001
From: Dimitri Diakopoulos <ddiakopoulos@gmail.com>
Date: Tue, 26 Jul 2016 10:29:59 -0700
Subject: [PATCH] hermite resampling just for fun

---
 examples/src/Main.cpp           | 10 ++++++----
 include/libnyquist/WavEncoder.h | 29 +++++++++++++++++++++++++++--
 2 files changed, 33 insertions(+), 6 deletions(-)

diff --git a/examples/src/Main.cpp b/examples/src/Main.cpp
index 2a31cad..a83a1b6 100644
--- a/examples/src/Main.cpp
+++ b/examples/src/Main.cpp
@@ -41,7 +41,7 @@ int main(int argc, const char **argv) try
 		//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/44100/32/test.wav");
 		//loader.Load(fileData.get(), "test_data/1ch/44100/64/test.wav");
 
 		// 2-channel wave
@@ -86,14 +86,16 @@ int main(int argc, const char **argv) try
 		//loader.Load(fileData.get(), "test_data/ad_hoc/44_16_mono.mpc");
 	}
 
+	/*
 	// Test Recording Capabilities of AudioDevice
-	fileData->samples.reserve(44100 * 2);
+	fileData->samples.reserve(44100 * 5);
 	fileData->channelCount = 1;
 	fileData->frameSize = 32;
-	fileData->lengthSeconds = 2.0;
+	fileData->lengthSeconds = 5.0;
 	fileData->sampleRate = 44100;
-	std::cout << "Starting recording for two seconds..." << std::endl;
+	std::cout << "Starting recording ..." << std::endl;
 	myDevice.Record(fileData->sampleRate * fileData->lengthSeconds, fileData->samples);
+	*/
 
 	// Libnyquist does not (currently) perform sample rate conversion - not exactly true, anymore. See below.
 	if (fileData->sampleRate != desiredSampleRate)
diff --git a/include/libnyquist/WavEncoder.h b/include/libnyquist/WavEncoder.h
index 8aedfb6..f66a03b 100644
--- a/include/libnyquist/WavEncoder.h
+++ b/include/libnyquist/WavEncoder.h
@@ -32,7 +32,6 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 
 namespace nqr
 {
-
 	// This is a naieve implementation of a resampling filter where a lerp is used as a bad low-pass.
 	// It very far from the ideal case and should be used with caution (or not at all) on signals that matter.
 	// It is included here to upsample 44.1k to 48k for the purposes of microphone input => Opus, where the the 
@@ -46,7 +45,7 @@ namespace nqr
 		{
 			uint32_t readIndex = static_cast<uint32_t>(virtualReadIndex);
 			i = virtualReadIndex - readIndex;
-			a = input[readIndex + 0];
+			a = input[readIndex + 0]; 
 			b = input[readIndex + 1];
 			sample = (1.0 - i) * a + i * b; // linear interpolate
 			output.push_back(sample);
@@ -54,6 +53,32 @@ namespace nqr
 		}
 	}
 
+	static inline double sample_hermite_4p_3o(double x, double * y)
+	{
+		static double c0, c1, c2, c3;
+		c0 = y[1];
+		c1 = (1.0/2.0)*(y[2]-y[0]);
+		c2 = (y[0] - (5.0/2.0)*y[1]) + (2.0*y[2] - (1.0/2.0)*y[3]);
+		c3 = (1.0/2.0)*(y[3]-y[0]) + (3.0/2.0)*(y[1]-y[2]);
+		return ((c3*x+c2)*x+c1)*x+c0;
+	}
+
+	static inline void hermite_resample(const double rate, const std::vector<float> & input, std::vector<float> & output, size_t samplesToProcess)
+	{
+		double virtualReadIndex = 1;
+		double i, sample;
+		uint32_t n = samplesToProcess - 1;
+		while (n--)
+		{
+			uint32_t readIndex = static_cast<uint32_t>(virtualReadIndex);
+			i = virtualReadIndex - readIndex;
+			double samps[4] = {input[readIndex - 1], input[readIndex], input[readIndex + 1], input[readIndex + 2]};
+			sample = sample_hermite_4p_3o(i, samps); // cubic hermite interpolate over 4 samples
+			output.push_back(sample);
+			virtualReadIndex += rate;
+		}
+	}
+
 	enum EncoderError
 	{
 		NoError,