Fixed rate change for multiple channel audio

waywardgeek · May 5, 2017 · 7b44193 · 7b44193
1 parent 6d65569
commit 7b44193
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Showing 4 changed files with 30 additions and 15 deletions.
diff --git a/Main.java b/Main.java
@@ -60,14 +60,14 @@ private static void runSonic(
     public static void main(
     	String[] argv) throws UnsupportedAudioFileException, IOException, LineUnavailableException
     {
-        float speed = 2.0f;
+        float speed = 1.0f;
         float pitch = 1.0f;
-        float rate = 1.0f;
+        float rate = 1.5f;
         float volume = 1.0f;
         boolean emulateChordPitch = false;
         int quality = 0;
 
-        AudioInputStream stream = AudioSystem.getAudioInputStream(new File("talking.wav"));
+        AudioInputStream stream = AudioSystem.getAudioInputStream(new File("stereo_test.wav"));
         AudioFormat format = stream.getFormat();
         int sampleRate = (int)format.getSampleRate();
         int numChannels = format.getChannels(); 

diff --git a/Sonic.java b/Sonic.java
@@ -736,12 +736,12 @@ private void adjustPitch(
     // Interpolate the new output sample.
     private short interpolate(
         short in[],
-        int inPos,
+        int inPos,  // Index to first sample which already includes channel offset.
         int oldSampleRate,
         int newSampleRate)
     {
-        short left = in[inPos*numChannels];
-        short right = in[inPos*numChannels + numChannels];
+        short left = in[inPos];
+        short right = in[inPos + numChannels];
         int position = newRatePosition*oldSampleRate;
         int leftPosition = oldRatePosition*newSampleRate;
         int rightPosition = (oldRatePosition + 1)*newSampleRate;
@@ -774,8 +774,8 @@ private void adjustRate(
             while((oldRatePosition + 1)*newSampleRate > newRatePosition*oldSampleRate) {
                 enlargeOutputBufferIfNeeded(1);
                 for(int i = 0; i < numChannels; i++) {
-                    outputBuffer[numOutputSamples*numChannels + i] = interpolate(pitchBuffer, position + i,
-                            oldSampleRate, newSampleRate);
+                    outputBuffer[numOutputSamples*numChannels + i] = interpolate(pitchBuffer,
+                            position*numChannels + i, oldSampleRate, newSampleRate);
                 }
                 newRatePosition++;
                 numOutputSamples++;

diff --git a/funcgen.c b/funcgen.c
@@ -6,17 +6,23 @@
 #define M_PI 3.14159265358979323846
 #endif
 
-#define SAMPLE_RATE 11025
-#define NUM_SAMPLES 11025
-#define NOISE 15
+#define SAMPLE_RATE 22050
+#define NUM_SAMPLES 22050
+#define NOISE 0
 
 /* Fundamental */
 #define MID_FREQ 150
 #define FREQ_CHANGE 50
+#define MAX_FREQ 2000.0
 
 /* Harmonics */
-#define NUM_HARMONICS 5
-static double harmonic[NUM_HARMONICS] = {4000.0, 4000.0, 2000.0, 0.0, 1000.0};
+#define NUM_HARMONICS 20
+static double harmonic[NUM_HARMONICS] = {1000.0, 1000.0, 1000.0, 1000.0, 1000.0, 1000.0, 1000.0, 1000.0, 1000.0, 1000.0,
+    1000.0, 1000.0, 1000.0, 1000.0, 1000.0, 1000.0, 1000.0, 1000.0, 1000.0, 1000.0};
+/* The phase seems to make no difference in how it sounds to me. */
+static double phase[NUM_HARMONICS] = {0.0, 1.0, 2.0, 5.5, 4.1234, 2.8, 6.0, 4.5, 2.67, 0.543,
+    0.01, 1.234, 2.345, 3.456, 5.9, 4.8, 0.89, 1.33, 3.8, 3.14159};
+/* static double phase[NUM_HARMONICS] = {0.0,}; */
 
 int main() {
     short samples[NUM_SAMPLES];
@@ -28,7 +34,16 @@ int main() {
         samples[i] = 0.0;
         int j;
         for (j = 0; j < NUM_HARMONICS; j++) {
-            samples[i] += harmonic[j]*sin((j+1)*w);
+            double amplitude = harmonic[j];
+            if ((j+1)*freq > MAX_FREQ) {
+                if ((j)*freq >= MAX_FREQ) {
+                    amplitude = 0.0;
+                } else {
+                    double ratio = ((j+1)*freq - MAX_FREQ)/freq;
+                    amplitude *= 1.0 - ratio;
+                }
+            }
+            samples[i] += amplitude*sin((j+1)*w + phase[j]);
         }
         /* Add noise */
         samples[i] += (float)NOISE*rand()/RAND_MAX;

diff --git a/sonic.c b/sonic.c
@@ -945,7 +945,7 @@ static int adjustRate(
                 return 0;
             }
             out = stream->outputBuffer + stream->numOutputSamples*numChannels;
-            in = stream->pitchBuffer + position;
+            in = stream->pitchBuffer + position*numChannels;
             for(i = 0; i < numChannels; i++) {
                 *out++ = interpolate(stream, in, oldSampleRate, newSampleRate);
                 in++;