Update

src/Home/Tracker/Lib/AC97.ZC

@@ -107,7 +107,7 @@ I64 AudioSFXPlay(U32 *data, I64 length)
 		return stream;
 	for (i = 0; i < length; i++)
 		FifoI64Ins(audio.output[stream].data, data[i]);
-	return stream;
+	return stream;
 }
 
 U0 AC97OutputMix(U32 *buf, I64 length = NULL)
@@ -188,6 +188,13 @@ I64 AC97Init()
 	OutU32(ac97.nabm + GLOBAL_CTL, 0x03);
 	OutU16(ac97.nam + RESET, 0xFFFF);
 
+	// FIXME: 
+	// i set the frequency to 44.1 manually
+	// DONT DO THIS!!!	
+	OutU16(ac97.nam + EXT_FRONT_RATE, 44100);
+
+
+
 	// Set PCM Output to Max volume
 	OutU16(ac97.nam + PCM_VOL, 0x0000);
 

src/Home/Tracker/Load.ZC

@@ -17,6 +17,8 @@ include_noreindex "Lib/ELF64";
 
 //include_noreindex "Lib/uPNG";
 // include_noreindex "Player";
+U16 currentRate = InU16(ac97.nam + EXT_FRONT_RATE);
+"currentRate: %d\n", currentRate;
 
 #include "Classes"
 // include_noreindex "MIDIHandling"

src/Home/Tracker/MusicTracker.ZC

@@ -2,9 +2,26 @@
 #include "WaveformGen"
 
 U0 AudioPlayNote(U8 note, U8 velocity, U8 instrument) {
-    U32 buffer[SAMPLE_RATE];
+
+	// ultimately, we should use AC97OutputMix to be multi channel
+
+    //U32 buffer[SAMPLE_RATE];
+
+	U32 *buffer;
+	I64 bufferSize;
+
+	if (instrument == SAMPLE)
+	{
+		bufferSize = 132300;
+	 	buffer = MAlloc(bufferSize * sizeof(U32));
+	}
+	else
+	{
+		bufferSize = SAMPLE_RATE;
+		buffer = MAlloc(bufferSize * sizeof(U32));
+	}
+
     F64 freq = MidiToFreq(note);
-    // GenerateSineWave(buffer, SAMPLE_RATE, freq, velocity);
 
     switch(instrument) {
         case PULSE1:
@@ -20,7 +37,7 @@ U0 AudioPlayNote(U8 note, U8 velocity, U8 instrument) {
             GenerateNoiseWave(buffer, SAMPLE_RATE, freq, velocity);
             break;
         case SAMPLE:
-            PlaySample(buffer, SAMPLE_RATE, velocity);
+            PlaySample(buffer, 132300, velocity);
             break;
         default:
             GenerateSineWave(buffer, SAMPLE_RATE, freq, velocity);
@@ -31,7 +48,28 @@ U0 AudioPlayNote(U8 note, U8 velocity, U8 instrument) {
 
     // Play the buffer using the AC97 driver:
     AudioSFXPlay(buffer, SAMPLE_RATE);
-	"Instrument: $$LTGREEN$$ %d $$DKGRAY$$|$$FG$$ Note: $$YELLOW$$%d$$FG$$\n", instrument, note;
+	U8 *instrument_name = "DEFAULT";
+	if (instrument == SAMPLE) instrument_name = "SAMPLE";
+	if (instrument == PULSE1) instrument_name = "PULSE1";
+	if (instrument == PULSE2) instrument_name = "PULSE2";
+	if (instrument == TRIANGLE) instrument_name = "TRIANGLE";
+	if (instrument == NOISE) instrument_name = "NOISE";
+	"Instrument: $$LTGREEN$$ %s $$DKGRAY$$|$$FG$$ Note: $$YELLOW$$%d$$FG$$\n", instrument_name, note;
+
+	Free(buffer);
+}
+
+U0 PlayPattern(Pattern *pattern) {
+    I64 row;
+    NoteCell *cell;
+    for (row = 0; row < TRACK_LENGTH; row++) {
+        cell = &pattern->cells[row];
+        if (cell->note) {
+            AudioPlayNote(cell->note, cell->velocity, cell->instrument);
+        }
+		//if (cell->instrument == SAMPLE) Sleep(2000);
+        Sleep(1000);  // Adjust for tempo
+    }
 }
 
 U0 EnterPattern(Pattern *pattern) {
@@ -48,21 +86,9 @@ U0 EnterPattern(Pattern *pattern) {
     }
 }
 
-U0 PlayPattern(Pattern *pattern) {
-    I64 row;
-    NoteCell *cell;
-    for (row = 0; row < TRACK_LENGTH; row++) {
-        cell = &pattern->cells[row];
-        if (cell->note) {
-            AudioPlayNote(cell->note, cell->velocity, cell->instrument);
-        }
-        Sleep(2900);  // Adjust for tempo
-    }
-}
-
 U0 MusicTracker() {
 
-	LoadSample("~/Tracker/Samples/Sample.WAV");
+	LoadSample("~/Tracker/Samples/OrchHit.WAV");
 
     Song song;
 

src/Home/Tracker/WaveformGen.ZC

@@ -39,7 +39,7 @@ U0 GeneratePulse2Wave(U32 *buffer, I64 length, F64 freq, U8 velocity) {
 	
 	for (i = 0; i < length; i++) {
 		I16 sample_value;
-		if (phase < 0.5 * PI) {
+		if (phase < 0.5 * PI) {
 			sample_value = -ToI64(amplitude);
 		} else {
 			sample_value = ToI64(amplitude);
@@ -132,7 +132,7 @@ U0 LoadSample(U8 *filename) {
     }
 
     gSampleSize = fileSize - sizeof(WAVHeader);
-    //gSampleData = MAlloc(gSampleSize);
+	//gSampleData = MAlloc(gSampleSize);
 
 
     //U8 *buffer = FileRead(filename, &fileSize);
@@ -162,6 +162,7 @@ U0 LoadSample(U8 *filename) {
 
 	// DEBUG
 	//gSampleSize = sizeof(fileContent) - 36;
+	//gSampleSize = fileSize - 36;
 	//U8 *audioDataStart = fileContent + 36; // Move pointer after header MANUAL 36 bytes
 	//MemCopy(gSampleData, audioDataStart, 36);
 
@@ -169,6 +170,7 @@ U0 LoadSample(U8 *filename) {
 	//Free(fileContent);
 
 	// test without skipping header
+
 	gSampleData = FileRead(filename, &gSampleSize);
 
     // Print some info about the sample
@@ -185,11 +187,47 @@ U0 PlaySample(U32 *buffer, I64 duration, U8 velocity) {
         return;
     }
     // Just as an example: copy loaded sample to buffer
-    // This step will depend on how you are processing and playing the audio
-    I64 samplesToCopy = Min(gSampleSize, duration); // don't overflow the buffer
+	I64 samplesToCopy = Min(gSampleSize, duration); // don't overflow the buffer
     MemCopy(buffer, gSampleData, samplesToCopy);
+    
+	//I64 numSamples = gSampleSize / 2;
+	//I64 numStereoPairs = numSamples / 2;
+	//I64 pairsToCopy = Min(numStereoPairs, duration);
 
-    // Here you'd typically send the buffer to your audio playing routine
+	//I64 i;
+
+	//for (i = 0; i < pairsToCopy; i++) {
+	//	I64 leftSample = (gSampleData[4*i] | (gSampleData[4*i + 1] << 8));
+	//	I64 rightSample = (gSampleData[4*i + 2] | (gSampleData[4*i + 3] << 8));
+	//	buffer[i] = (leftSample << 16) | (rightSample & 0xFFFF);
+	//}
+
+
+	// speed is almost good but right ear only...
+	//I64 numSamples = gSampleSize / 2;
+	//I64 samplesToCopy = Min(numSamples, duration); // don't overflow the buffer
+	//I64 i;
+
+	//for (i=0; i < samplesToCopy; i++)
+	//{
+	//	I64 sample = (gSampleData[2 * i] | (gSampleData[2 *i + 1] << 8));
+	//	buffer[i] = sample << 16;
+	//}
+	
+
+
+    //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
+    //   }
+    //}
 }
 
 // I64 sample_rate = SAMPLE_RATE // whatever your sample rate is