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https://github.com/Zeal-Operating-System/ZealOS.git
synced 2024-12-25 23:10:32 +00:00
Sampler fix
This commit is contained in:
y4my4my4m
parent
db77fc64d9
commit
ecdb2d01c0
3 changed files with 166 additions and 89 deletions
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@ -12,7 +12,7 @@ class NoteCell {
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U8 effect; // Future expansion for effects
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};
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#define TRACK_LENGTH 10 // Length of each pattern in rows
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#define TRACK_LENGTH 13 // Length of each pattern in rows
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class Pattern {
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NoteCell cells[TRACK_LENGTH];
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};
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@ -24,6 +24,21 @@ class Song {
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};
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class WAVHeader {
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U8 riff[4]; // RIFF string
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U32 overall_size; // overall size of file in bytes
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U8 wave[4]; // WAVE string
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U8 fmt_chunk[4]; // fmt string with trailing null char
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U32 length; // length of format data. Should be 16 for PCM
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U16 format_type; // format type. 1 for PCM
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U16 channels; // number of channels
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U32 sample_rate; // sampling rate (blocks per second)
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U32 byterate; // SampleRate * NumChannels * BitsPerSample/8
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U16 block_align; // channels * bits/sample / 8
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U16 bits_per_sample;// bits per sample, 8- 8bits, 16- 16 bits etc
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};
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#define SAMPLE_RATE 44100
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#define PI 3.1415926535897932 // i know i can just use the pi symbol
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@ -4,7 +4,10 @@
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U0 AudioPlayNote(U8 note, U8 velocity, U8 instrument) {
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// ultimately, we should use AC97OutputMix to be multi channel
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FifoI64Flush(audio.output[0]);
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FifoI64Flush(audio.output[1]);
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FifoI64Flush(audio.output[2]);
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FifoI64Flush(audio.output[3]);
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//U32 buffer[SAMPLE_RATE];
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U32 *buffer;
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@ -12,12 +15,13 @@ U0 AudioPlayNote(U8 note, U8 velocity, U8 instrument) {
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if (instrument == SAMPLE)
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{
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bufferSize = 132300;
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//bufferSize = SAMPLE_RATE * 14 * 2; // 7sec * 2 two channels * 2 for some reason?
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bufferSize = 176400; // 1sec
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buffer = MAlloc(bufferSize * sizeof(U32));
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}
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else
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{
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bufferSize = SAMPLE_RATE;
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bufferSize = 11025; // SAMPLE_RATE change duration
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buffer = MAlloc(bufferSize * sizeof(U32));
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}
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@ -25,37 +29,42 @@ U0 AudioPlayNote(U8 note, U8 velocity, U8 instrument) {
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switch(instrument) {
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case PULSE1:
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GeneratePulse1Wave(buffer, SAMPLE_RATE, freq, velocity);
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GeneratePulse1Wave(buffer, bufferSize, freq, velocity);
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break;
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case PULSE2:
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GeneratePulse2Wave(buffer, SAMPLE_RATE, freq, velocity);
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GeneratePulse2Wave(buffer, bufferSize, freq, velocity);
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break;
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case TRIANGLE:
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GenerateTriangleWave(buffer, SAMPLE_RATE, freq, velocity);
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GenerateTriangleWave(buffer, bufferSize, freq, velocity);
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break;
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case NOISE:
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GenerateNoiseWave(buffer, SAMPLE_RATE, freq, velocity);
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GenerateNoiseWave(buffer, bufferSize, freq, velocity);
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break;
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case SAMPLE:
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PlaySample(buffer, 132300, velocity);
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PlaySample(buffer, bufferSize, note, velocity);
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Print("Buffer first values: %d, %d, %d, %d...\n", buffer[0], buffer[1], buffer[2], buffer[3]);
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break;
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default:
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GenerateSineWave(buffer, SAMPLE_RATE, freq, velocity);
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GenerateSineWave(buffer, bufferSize, freq, velocity);
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break;
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}
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//ApplyEnvelope(buffer, SAMPLE_RATE); // TODO: Apply fade in and fade out
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// Play the buffer using the AC97 driver:
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AudioSFXPlay(buffer, SAMPLE_RATE);
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U8 *instrument_name = "DEFAULT";
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if (instrument == SAMPLE) instrument_name = "SAMPLE";
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if (instrument == PULSE1) instrument_name = "PULSE1";
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if (instrument == PULSE2) instrument_name = "PULSE2";
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if (instrument == TRIANGLE) instrument_name = "TRIANGLE";
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if (instrument == NOISE) instrument_name = "NOISE";
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"Instrument: $$LTGREEN$$ %s $$DKGRAY$$|$$FG$$ Note: $$YELLOW$$%d$$FG$$\n", instrument_name, note;
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AudioSFXPlay(buffer, bufferSize);
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// silly debug print
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if (instrument == SAMPLE)
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"Instrument: $$RED$$ SAMPLE $$DKGRAY$$|$$FG$$ Note: $$YELLOW$$%d$$FG$$\n", note;
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if (instrument == PULSE1)
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"Instrument: $$GREEN$$ PULSE1 $$DKGRAY$$|$$FG$$ Note: $$YELLOW$$%d$$FG$$\n", note;
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if (instrument == PULSE2)
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"Instrument: $$CYAN$$ PULSE2 $$DKGRAY$$|$$FG$$ Note: $$YELLOW$$%d$$FG$$\n", note;
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if (instrument == TRIANGLE)
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"Instrument: $$YELLOW$$ TRIANGLE $$DKGRAY$$|$$FG$$ Note: $$YELLOW$$%d$$FG$$\n", note;
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if (instrument == NOISE)
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"Instrument: $$BROWN$$ NOISE $$DKGRAY$$|$$FG$$ Note: $$YELLOW$$%d$$FG$$\n", note;
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Free(buffer);
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}
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@ -67,8 +76,9 @@ U0 PlayPattern(Pattern *pattern) {
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if (cell->note) {
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AudioPlayNote(cell->note, cell->velocity, cell->instrument);
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}
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//if (cell->instrument == SAMPLE) Sleep(2000);
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Sleep(1000); // Adjust for tempo
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if (cell->instrument == SAMPLE) Sleep(1000); // wait for 6sec for the 7sec sample test
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//Sleep(1200); // Adjust for tempo
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Sleep(300);
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}
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}
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@ -93,47 +103,57 @@ U0 MusicTracker() {
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Song song;
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song.patterns[0].cells[0].note = 60; // C3
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song.patterns[0].cells[0].note = 60;
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song.patterns[0].cells[0].velocity = 100;
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song.patterns[0].cells[0].instrument = SAMPLE;
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song.patterns[0].cells[1].note = 60; // C3
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song.patterns[0].cells[1].note = 48; // C3
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song.patterns[0].cells[1].velocity = 100;
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song.patterns[0].cells[1].instrument = PULSE1;
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song.patterns[0].cells[1].instrument = PULSE2;
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song.patterns[0].cells[2].note = 62; // D
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song.patterns[0].cells[2].note = 60; // C3
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song.patterns[0].cells[2].velocity = 100;
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song.patterns[0].cells[2].instrument = PULSE1;
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song.patterns[0].cells[3].note = 64; // E
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song.patterns[0].cells[3].note = 62; // D
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song.patterns[0].cells[3].velocity = 100;
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song.patterns[0].cells[3].instrument = PULSE1;
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song.patterns[0].cells[4].note = 60; // C
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song.patterns[0].cells[4].note = 64; // E
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song.patterns[0].cells[4].velocity = 100;
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song.patterns[0].cells[4].instrument = PULSE2;
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song.patterns[0].cells[4].instrument = SAMPLE;
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song.patterns[0].cells[5].note = 62; // D
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song.patterns[0].cells[5].note = 65; // C
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song.patterns[0].cells[5].velocity = 100;
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song.patterns[0].cells[5].instrument = PULSE2;
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song.patterns[0].cells[6].note = 60; // C
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song.patterns[0].cells[6].note = 68; // D
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song.patterns[0].cells[6].velocity = 100;
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song.patterns[0].cells[6].instrument = TRIANGLE;
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song.patterns[0].cells[6].instrument = PULSE2;
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song.patterns[0].cells[7].note = 62; // D
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song.patterns[0].cells[7].note = 72; // C
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song.patterns[0].cells[7].velocity = 100;
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song.patterns[0].cells[7].instrument = TRIANGLE;
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song.patterns[0].cells[8].note = 64; // E
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song.patterns[0].cells[8].note = 48; // D
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song.patterns[0].cells[8].velocity = 100;
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song.patterns[0].cells[8].instrument = TRIANGLE;
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song.patterns[0].cells[9].note = 102;
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song.patterns[0].cells[9].note = 64; // E
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song.patterns[0].cells[9].velocity = 100;
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song.patterns[0].cells[9].instrument = NOISE;
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song.patterns[0].cells[9].instrument = TRIANGLE;
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song.patterns[0].cells[10].note = 102;
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song.patterns[0].cells[10].velocity = 70;
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song.patterns[0].cells[10].instrument = NOISE;
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song.patterns[0].cells[11].note = 125;
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song.patterns[0].cells[11].velocity = 35;
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song.patterns[0].cells[11].instrument = NOISE;
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song.patterns[0].cells[12].note = 65;
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song.patterns[0].cells[12].velocity = 80;
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song.patterns[0].cells[12].instrument = NOISE;
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// Clear(&song);
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I64 sc;
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@ -105,33 +105,18 @@ U0 ApplyEnvelope(U32 *buffer, I64 length) {
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// }
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// }
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class WAVHeader {
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U8 riff[4]; // RIFF string
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U32 overall_size; // overall size of file in bytes
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U8 wave[4]; // WAVE string
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U8 fmt_chunk[4]; // fmt string with trailing null char
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U32 length; // length of format data. Should be 16 for PCM
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U16 format_type; // format type. 1 for PCM
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U16 channels; // number of channels
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U32 sample_rate; // sampling rate (blocks per second)
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U32 byterate; // SampleRate * NumChannels * BitsPerSample/8
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U16 block_align; // channels * bits/sample / 8
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U16 bits_per_sample;// bits per sample, 8- 8bits, 16- 16 bits etc
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};
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U0 LoadSample(U8 *filename) {
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//WAVHeader header;
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I64 fileSize;
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// Free any previously loaded sample
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if (gSampleData) {
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Free(gSampleData);
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}
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// if (gSampleData) {
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// Free(gSampleData);
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// }
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gSampleSize = fileSize - sizeof(WAVHeader);
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//gSampleData = MAlloc(gSampleSize);
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@ -154,17 +139,17 @@ U0 LoadSample(U8 *filename) {
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//}
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//U8 *fileContent = FileRead(filename, &fileSize);
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//U8 *fileContent = FileRead(filename, &gSampleSize);
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//U8 *audioDataStart = fileContent + sizeof(WAVHeader); // Move pointer after header
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//MemCopy(gSampleData, audioDataStart, gSampleSize);
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// DEBUG
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//gSampleSize = sizeof(fileContent) - 36;
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//gSampleSize = fileSize - 36;
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//U8 *audioDataStart = fileContent + 36; // Move pointer after header MANUAL 36 bytes
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//MemCopy(gSampleData, audioDataStart, 36);
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// gSampleSize = fileSize - 44;
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//U8 *audioDataStart = fileContent + 44; // Move pointer after header MANUAL 36 bytes
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//MemCopy(gSampleData, audioDataStart, gSampleSize);
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//Free(fileContent);
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@ -180,42 +165,95 @@ 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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// }
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U0 PlaySample(U32 *buffer, I64 duration, U8 velocity) {
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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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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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// Just as an example: copy loaded sample to 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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// F64 multiplier = GetPlaybackRateMultiplier(note, C3_NOTE_VALUE);
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// Print("Playing with multiplier: %f\n", multiplier); // Add this print statement
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// I64 srcIndex = 0;
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// I64 destIndex;
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// for (destIndex = 0; destIndex < duration; destIndex++) {
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// if (srcIndex < gSampleSize) {
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// buffer[destIndex] = gSampleData[srcIndex];
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// srcIndex = ToI64(srcIndex + multiplier);
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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); // Print the last value of srcIndex after the loop
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// Start after the WAV header
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// I64 destIndex;
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// F64 srcPosition = 44.0; // Start after WAV header
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// for (destIndex = 0; destIndex < duration; destIndex++) {
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// I64 baseIndex = ToI64(srcPosition);
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// if (baseIndex < gSampleSize - 8) { // Ensure there are enough samples ahead for interpolation
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// // Read the two stereo samples we're interpolating between
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// I16 leftSample1 = (gSampleData[baseIndex + 1] << 8) | gSampleData[baseIndex];
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// I16 rightSample1 = (gSampleData[baseIndex + 3] << 8) | gSampleData[baseIndex + 2];
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// I16 leftSample2 = (gSampleData[baseIndex + 5] << 8) | gSampleData[baseIndex + 4];
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// I16 rightSample2 = (gSampleData[baseIndex + 7] << 8) | gSampleData[baseIndex + 6];
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// F64 fraction = srcPosition - baseIndex;
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// // Linearly interpolate between the two samples for each channel
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// I16 leftMixed = ClampI16((1 - fraction) * leftSample1 + fraction * leftSample2);
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// I16 rightMixed = ClampI16((1 - fraction) * rightSample1 + fraction * rightSample2);
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// buffer[destIndex] = (leftMixed & 0xFFFF) | ((rightMixed & 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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// srcPosition += 4.0 * multiplier; // Move by one stereo sample, adjusted by the multiplier
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// }
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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 * 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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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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Print("Last srcIndex: %d\n", srcIndex);
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//I64 numSamples = gSampleSize / 2;
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//I64 numStereoPairs = numSamples / 2;
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//I64 pairsToCopy = Min(numStereoPairs, duration);
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//I64 i;
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//for (i = 0; i < pairsToCopy; i++) {
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// I64 leftSample = (gSampleData[4*i] | (gSampleData[4*i + 1] << 8));
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// I64 rightSample = (gSampleData[4*i + 2] | (gSampleData[4*i + 3] << 8));
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// buffer[i] = (leftSample << 16) | (rightSample & 0xFFFF);
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//}
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// speed is almost good but right ear only...
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//I64 numSamples = gSampleSize / 2;
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//I64 samplesToCopy = Min(numSamples, duration); // don't overflow the buffer
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//I64 i;
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//for (i=0; i < samplesToCopy; i++)
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//{
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// I64 sample = (gSampleData[2 * i] | (gSampleData[2 *i + 1] << 8));
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// buffer[i] = sample << 16;
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//}
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//F64 playbackStep = 1.0; // 1.0 means original speed. <1.0 is slower, >1.0 is faster.
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//F64 playbackPointer = 0.0;
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//I64 i;
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@ -228,6 +266,10 @@ U0 PlaySample(U32 *buffer, I64 duration, U8 velocity) {
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// buffer[i] = 0; // or whatever value represents silence in your buffer
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// }
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//}
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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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