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interpolated pitch shfting

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y4my4my4m 2023-09-04 14:36:05 +09:00
parent ecdb2d01c0
commit b34fe94d08
1 changed files with 82 additions and 73 deletions
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155
src/Home/Tracker/WaveformGen.ZC
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@ -178,95 +178,104 @@ F64 GetPlaybackRateMultiplier(U8 targetNote, U8 referenceNote) {
return Pow(2.0, semitoneDifference / 12.0); return Pow(2.0, semitoneDifference / 12.0);
} }
#define WINDOW_SIZE 10
F64 sinc(F64 x) {
if (x == 0.0) {
return 1.0;
} else {
return sin(PI * x) / (PI * x);
}
}
I16 RoundF64(F64 val) {
if (val < 0.0) {
return val - 0.5;
} else {
return val + 0.5;
}
}
I16 ClampToI16(I64 value) {
if (value > 32767) return 32767;
if (value < -32768) return -32768;
return value;
}
I64 ConvertU8PairToI64(U8 msb, U8 lsb) {
I64 val = (msb << 8) | lsb;
if (val & 0x8000) {
val |= 0xFFFF0000; // sign extend if negative
}
return val;
}
I16 WindowedSincInterpolation(F64 position) {
I64 baseIndex = ToI64(position);
F64 fraction = position - baseIndex;
F64 result = 0.0;
I64 i;
for (i = -WINDOW_SIZE; i <= WINDOW_SIZE; i++) {
F64 sample;
if (baseIndex + i >= 0 && baseIndex + i < gSampleSize) {
sample = ConvertU8ToI16(gSampleData[2 * (baseIndex + i)], gSampleData[2 * (baseIndex + i) + 1]);
} else {
sample = 0.0;
}
result += sample * sinc(i - fraction) * 0.54 - 0.46 * cos(2.0 * PI * (i - fraction) / (2 * WINDOW_SIZE + 1));
}
return ClampToI16(RoundF64(result));
}
U0 PlaySample(U32 *buffer, I64 duration, U8 note, U8 velocity) { U0 PlaySample(U32 *buffer, I64 duration, U8 note, U8 velocity) {
if (!gSampleData || !gSampleSize) { if (!gSampleData || !gSampleSize) {
Print("Sample not loaded.\n"); Print("Sample not loaded.\n");
return; return;
} }
// F64 multiplier = GetPlaybackRateMultiplier(note, C3_NOTE_VALUE);
// Print("Playing with multiplier: %f\n", multiplier); // Add this print statement
// I64 srcIndex = 0; F64 multiplier = GetPlaybackRateMultiplier(playedNote, 60);
// I64 destIndex; Print("multiplier: %f\n", multiplier);
// for (destIndex = 0; destIndex < duration; destIndex++) {
// if (srcIndex < gSampleSize) {
// buffer[destIndex] = gSampleData[srcIndex];
// srcIndex = ToI64(srcIndex + multiplier);
// } else {
// buffer[destIndex] = 0; // fill the rest with silence
// }
// }
// Print("Last srcIndex: %d\n", srcIndex); // Print the last value of srcIndex after the loop
// Start after the WAV header
// I64 destIndex;
// F64 srcPosition = 44.0; // Start after WAV header
// for (destIndex = 0; destIndex < duration; destIndex++) {
// I64 baseIndex = ToI64(srcPosition);
// if (baseIndex < gSampleSize - 8) { // Ensure there are enough samples ahead for interpolation
// // Read the two stereo samples we're interpolating between
// I16 leftSample1 = (gSampleData[baseIndex + 1] << 8) | gSampleData[baseIndex];
// I16 rightSample1 = (gSampleData[baseIndex + 3] << 8) | gSampleData[baseIndex + 2];
// I16 leftSample2 = (gSampleData[baseIndex + 5] << 8) | gSampleData[baseIndex + 4];
// I16 rightSample2 = (gSampleData[baseIndex + 7] << 8) | gSampleData[baseIndex + 6];
// F64 fraction = srcPosition - baseIndex;
// // Linearly interpolate between the two samples for each channel
// I16 leftMixed = ClampI16((1 - fraction) * leftSample1 + fraction * leftSample2);
// I16 rightMixed = ClampI16((1 - fraction) * rightSample1 + fraction * rightSample2);
// buffer[destIndex] = (leftMixed & 0xFFFF) | ((rightMixed & 0xFFFF) << 16);
// } else {
// buffer[destIndex] = 0; // fill the rest with silence
// }
// srcPosition += 4.0 * multiplier; // Move by one stereo sample, adjusted by the multiplier
// }
I64 destIndex; I64 destIndex;
F64 srcIndex = 44.0; // Start after WAV header F64 srcIndex = 44.0; // Start after WAV header
for (destIndex = 0; destIndex < duration; destIndex++) { for (destIndex = 0; destIndex < duration; destIndex++) {
F64 realIndex = srcIndex + destIndex * multiplier * 4; F64 realIndex = srcIndex + destIndex * multiplier;
I64 baseIndex = ToI64(realIndex);
F64 fraction = realIndex - baseIndex; I16 sample_value = WindowedSincInterpolation(realIndex);
buffer[destIndex] = (sample_value << 16) | (sample_value & 0xFFFF);
}
if (baseIndex < gSampleSize - 8) { // Ensure we can access two stereo samples
U32 leftSample1 = gSampleData[baseIndex] + (gSampleData[baseIndex + 1] << 8);
U32 rightSample1 = gSampleData[baseIndex + 2] + (gSampleData[baseIndex + 3] << 8);
U32 leftSample2 = gSampleData[baseIndex + 4] + (gSampleData[baseIndex + 5] << 8); // I64 destIndex;
U32 rightSample2 = gSampleData[baseIndex + 6] + (gSampleData[baseIndex + 7] << 8); // F64 srcIndex = 44.0; // Start after WAV header
// Linear interpolation // for (destIndex = 0; destIndex < duration; destIndex++) {
U32 leftSample = leftSample1 + ((leftSample2 - leftSample1) * fraction); // F64 realIndex = srcIndex + destIndex * multiplier * 4;
U32 rightSample = rightSample1 + ((rightSample2 - rightSample1) * fraction); // I64 baseIndex = ToI64(realIndex);
// F64 fraction = realIndex - baseIndex;
buffer[destIndex] = (leftSample & 0xFFFF) | ((rightSample & 0xFFFF) << 16); // if (baseIndex < gSampleSize - 8) { // Ensure we can access two stereo samples
} else { // U32 leftSample1 = gSampleData[baseIndex] + (gSampleData[baseIndex + 1] << 8);
buffer[destIndex] = 0; // fill the rest with silence // U32 rightSample1 = gSampleData[baseIndex + 2] + (gSampleData[baseIndex + 3] << 8);
}
} // U32 leftSample2 = gSampleData[baseIndex + 4] + (gSampleData[baseIndex + 5] << 8);
Print("Last srcIndex: %d\n", srcIndex); // U32 rightSample2 = gSampleData[baseIndex + 6] + (gSampleData[baseIndex + 7] << 8);
// // Linear interpolation
// U32 leftSample = leftSample1 + ((leftSample2 - leftSample1) * fraction);
// U32 rightSample = rightSample1 + ((rightSample2 - rightSample1) * fraction);
// buffer[destIndex] = (leftSample & 0xFFFF) | ((rightSample & 0xFFFF) << 16);
// } else {
// buffer[destIndex] = 0; // fill the rest with silence
// }
// }
// Print("Last srcIndex: %d\n", srcIndex);
//F64 playbackStep = 1.0; // 1.0 means original speed. <1.0 is slower, >1.0 is faster.
//F64 playbackPointer = 0.0;
//I64 i;
//for (i = 0; i < duration; i++) {
// if (playbackPointer < gSampleSize) {
// buffer[i] = gSampleData[ToI64(playbackPointer)]; // get the sample at the rounded-down playback pointer
// playbackPointer += playbackStep;
// } else {
// buffer[i] = 0; // or whatever value represents silence in your buffer
// }
//}
// Simply play the buffer // Simply play the buffer
//I64 samplesToCopy = Min(gSampleSize, duration); // don't overflow the buffer //I64 samplesToCopy = Min(gSampleSize, duration); // don't overflow the buffer
//MemCopy(buffer, gSampleData, samplesToCopy); //MemCopy(buffer, gSampleData, samplesToCopy);