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https://github.com/Zeal-Operating-System/ZealOS.git
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revert to non-interpolated and enter pattern
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y4my4my4m
parent
b34fe94d08
commit
ed4f56bca5
2 changed files with 92 additions and 99 deletions
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@ -83,16 +83,24 @@ U0 PlayPattern(Pattern *pattern) {
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}
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U0 EnterPattern(Pattern *pattern) {
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I64 row;
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I64 row, sc;
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NoteCell *cell;
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for (row = 0; row < TRACK_LENGTH; row++) {
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cell = &pattern->cells[row];
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Print("Enter note for row %d (0-127, 0 for none): ", row);
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// cell->note = InU8;
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// if (cell->note) {
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// Print("Enter velocity for note (1-127): ");
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// cell->velocity = InU8;
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// }
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cell->note = KeyGet(&sc);
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"%d\n", cell->note;
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if (cell->note) {
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Print("Enter velocity for note (1-127): ");
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cell->velocity = KeyGet(&sc);
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"%d\n", cell->velocity;
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if (cell->velocity) {
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Print("Enter instrument for note (0-4): ");
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cell->instrument = KeyGet(&sc);
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"%d\n", cell->instrument;
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AudioPlayNote(cell->note, cell->velocity, cell->instrument);
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}
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}
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}
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}
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@ -165,124 +165,109 @@ U0 LoadSample(U8 *filename) {
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}
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// // Clamping function without casting
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// I16 ClampI16(I16 value) {
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// if (value < -32768) return -32768;
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// if (value > 32767) return 32767;
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// return value;
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// Interpolation (works? but slow and distorted)
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// #define WINDOW_SIZE 10
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// F64 Sinc(F64 x) {
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// if (x == 0.0) {
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// return 1.0;
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// } else {
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// return Sin(PI * x) / (PI * x);
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// }
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// }
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// I16 RoundF64(F64 val) {
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// if (val < 0.0) {
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// return val - 0.5;
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// } else {
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// return val + 0.5;
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// }
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// }
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// I16 ClampToI16(I64 value) {
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// if (value > 32767) return 32767;
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// if (value < -32768) return -32768;
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// return value;
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// }
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// I64 ConvertU8PairToI64(U8 msb, U8 lsb) {
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// I64 val = (msb << 8) | lsb;
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// if (val & 0x8000) {
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// val |= 0xFFFF0000; // sign extend if negative
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// }
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// return val;
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// }
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// I16 ConvertU8ToI16(U8 lowByte, U8 highByte) {
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// I16 result = highByte;
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// result = (result << 8) | lowByte;
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// return result;
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// }
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// I16 WindowedSincInterpolation(F64 position) {
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// I64 baseIndex = ToI64(position);
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// F64 fraction = position - baseIndex;
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// F64 result = 0.0;
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// I64 i;
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// for (i = -WINDOW_SIZE; i <= WINDOW_SIZE; i++) {
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// F64 sample;
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// if (baseIndex + i >= 0 && baseIndex + i < gSampleSize) {
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// sample = ConvertU8ToI16(gSampleData[2 * (baseIndex + i)], gSampleData[2 * (baseIndex + i) + 1]);
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// } else {
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// sample = 0.0;
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// }
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// result += sample * Sinc(i - fraction) * 0.54 - 0.46 * Cos(2.0 * PI * (i - fraction) / (2 * WINDOW_SIZE + 1));
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// }
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// return ClampToI16(RoundF64(result));
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// }
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F64 RoundToNearestHalf(F64 value) {
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return Round(value * 2.0) / 2.0;
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}
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F64 GetPlaybackRateMultiplier(U8 targetNote, U8 referenceNote) {
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I64 semitoneDifference = targetNote - referenceNote;
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return Pow(2.0, semitoneDifference / 12.0);
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}
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#define WINDOW_SIZE 10
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F64 sinc(F64 x) {
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if (x == 0.0) {
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return 1.0;
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} else {
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return sin(PI * x) / (PI * x);
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}
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}
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I16 RoundF64(F64 val) {
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if (val < 0.0) {
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return val - 0.5;
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} else {
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return val + 0.5;
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}
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}
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I16 ClampToI16(I64 value) {
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if (value > 32767) return 32767;
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if (value < -32768) return -32768;
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return value;
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}
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I64 ConvertU8PairToI64(U8 msb, U8 lsb) {
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I64 val = (msb << 8) | lsb;
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if (val & 0x8000) {
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val |= 0xFFFF0000; // sign extend if negative
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}
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return val;
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}
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I16 WindowedSincInterpolation(F64 position) {
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I64 baseIndex = ToI64(position);
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F64 fraction = position - baseIndex;
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F64 result = 0.0;
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I64 i;
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for (i = -WINDOW_SIZE; i <= WINDOW_SIZE; i++) {
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F64 sample;
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if (baseIndex + i >= 0 && baseIndex + i < gSampleSize) {
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sample = ConvertU8ToI16(gSampleData[2 * (baseIndex + i)], gSampleData[2 * (baseIndex + i) + 1]);
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} else {
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sample = 0.0;
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}
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result += sample * sinc(i - fraction) * 0.54 - 0.46 * cos(2.0 * PI * (i - fraction) / (2 * WINDOW_SIZE + 1));
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}
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return ClampToI16(RoundF64(result));
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}
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U0 PlaySample(U32 *buffer, I64 duration, U8 note, U8 velocity) {
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if (!gSampleData || !gSampleSize) {
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Print("Sample not loaded.\n");
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return;
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}
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F64 multiplier = GetPlaybackRateMultiplier(playedNote, 60);
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Print("multiplier: %f\n", multiplier);
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F64 multiplier = GetPlaybackRateMultiplier(note, 60);
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multiplier = RoundToNearestHalf(multiplier);
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I64 destIndex;
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F64 srcIndex = 44.0; // Start after WAV header
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for (destIndex = 0; destIndex < duration; destIndex++) {
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F64 realIndex = srcIndex + destIndex * multiplier;
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I16 sample_value = WindowedSincInterpolation(realIndex);
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buffer[destIndex] = (sample_value << 16) | (sample_value & 0xFFFF);
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}
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for (destIndex = 0; destIndex < duration; destIndex++) {
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F64 realIndex = srcIndex + destIndex * multiplier * 4;
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I64 baseIndex = ToI64(realIndex);
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F64 fraction = realIndex - baseIndex;
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if (baseIndex < gSampleSize - 8) { // Ensure we can access two stereo samples
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U32 leftSample1 = gSampleData[baseIndex] + (gSampleData[baseIndex + 1] << 8);
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U32 rightSample1 = gSampleData[baseIndex + 2] + (gSampleData[baseIndex + 3] << 8);
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// I64 destIndex;
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// F64 srcIndex = 44.0; // Start after WAV header
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U32 leftSample2 = gSampleData[baseIndex + 4] + (gSampleData[baseIndex + 5] << 8);
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U32 rightSample2 = gSampleData[baseIndex + 6] + (gSampleData[baseIndex + 7] << 8);
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// for (destIndex = 0; destIndex < duration; destIndex++) {
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// F64 realIndex = srcIndex + destIndex * multiplier * 4;
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// I64 baseIndex = ToI64(realIndex);
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// F64 fraction = realIndex - baseIndex;
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// Linear interpolation
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U32 leftSample = leftSample1 + ((leftSample2 - leftSample1) * fraction);
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U32 rightSample = rightSample1 + ((rightSample2 - rightSample1) * fraction);
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// if (baseIndex < gSampleSize - 8) { // Ensure we can access two stereo samples
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// U32 leftSample1 = gSampleData[baseIndex] + (gSampleData[baseIndex + 1] << 8);
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// U32 rightSample1 = gSampleData[baseIndex + 2] + (gSampleData[baseIndex + 3] << 8);
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// U32 leftSample2 = gSampleData[baseIndex + 4] + (gSampleData[baseIndex + 5] << 8);
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// U32 rightSample2 = gSampleData[baseIndex + 6] + (gSampleData[baseIndex + 7] << 8);
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// // Linear interpolation
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// U32 leftSample = leftSample1 + ((leftSample2 - leftSample1) * fraction);
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// U32 rightSample = rightSample1 + ((rightSample2 - rightSample1) * fraction);
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// buffer[destIndex] = (leftSample & 0xFFFF) | ((rightSample & 0xFFFF) << 16);
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// } else {
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// buffer[destIndex] = 0; // fill the rest with silence
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// }
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// }
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buffer[destIndex] = (leftSample & 0xFFFF) | ((rightSample & 0xFFFF) << 16);
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} else {
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buffer[destIndex] = 0; // fill the rest with silence
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}
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}
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// Print("Last srcIndex: %d\n", srcIndex);
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// Simply play the buffer
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//I64 samplesToCopy = Min(gSampleSize, duration); // don't overflow the buffer
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//MemCopy(buffer, gSampleData, samplesToCopy);
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}
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// I64 sample_rate = SAMPLE_RATE // whatever your sample rate is
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// for (I64 i = 0; i < sample_duration * sample_rate; i++) {
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// sample = sin(2.0 * PI * freq * i / sample_rate);
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// // Then send 'sample' to your audio buffer/output.
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// }
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