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More progress on audio and rendering

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This commit is contained in:
Bruno Rybársky
2025-06-02 18:04:15 +02:00
parent 84805b92cb
commit 0c3e2aa730
14 changed files with 569 additions and 264 deletions
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213
util/audio.c
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@@ -6,6 +6,11 @@
AudioData audioData;
#define MAX_MIDI_EVENTS 1024
MidiEvent midiEvents[MAX_MIDI_EVENTS];
int midiEventCount = 0;
int nextMidiEvent = 0;
uint16_t getAvailableChannel() {
for (uint16_t i = 0; i < NUM_SYNTH_VOICES; i++) {
if (audioData.synthVoices[i].volume == 0) {
@@ -39,93 +44,209 @@ static void compute_stereo_gains(float pan, float *outL, float *outR) {
// e.g. *outL *= 0.7071f; *outR *= 0.7071f;
}
// This callback now writes stereo frames: interleaved L/R floats.
// Improved audio callback with anti-clipping and smooth fade-out
void audio_callback(void *userdata, Uint8 *stream, int len) {
AudioData *audio = (AudioData *) userdata;
// 'len' is total bytes; each sample‐frame is 2 floats (L+R), i.e. 2 * sizeof(float).
int frames = len / (2 * sizeof(float));
int frames = len / (2 * sizeof(float)); // Stereo frame count
float elapsedSec = audio->totalSamples / SAMPLE_RATE;
audio->totalSamples += frames;
while (nextMidiEvent < midiEventCount &&
midiEvents[nextMidiEvent].timeSec <= elapsedSec) {
MidiEvent *ev = &midiEvents[nextMidiEvent];
if (ev->type == 0 && ev->velocity > 0) {
// Note On
for (int i = NUM_SYNTH_VOICES - 4; i < NUM_SYNTH_VOICES; ++i) {
SynthVoice *v = &audio->synthVoices[i];
if (v->volume == 0) {
float freq = 440.0f * powf(2.0f, (ev->note - 69) / 12.0f);
v->frequency = (uint16_t) freq;
v->volume = ev->velocity * 2;
v->waveform = WAVE_SQUARE;
v->smoothedAmp = 0;
break;
}
}
} else {
// Note Off
for (int i = NUM_SYNTH_VOICES - 4; i < NUM_SYNTH_VOICES; ++i) {
SynthVoice *v = &audio->synthVoices[i];
float freq = 440.0f * powf(2.0f, (ev->note - 69) / 12.0f);
if ((uint16_t)freq == v->frequency) {
v->volume = 0;
}
}
}
nextMidiEvent++;
}
// Zero out the entire output buffer (silence)
// We’ll accumulate into it.
// Each float is 4 bytes, so total floats = 2 * frames.
float *outBuf = (float *) stream;
for (int i = 0; i < 2 * frames; ++i) {
outBuf[i] = 0.0f;
}
// Precompute the listener center
float listenerCx = audio->playerRect->x + audio->playerRect->w * 0.5f;
// For each synth voice, mix into the stereo buffer
int *voiceCounts = calloc(frames, sizeof(int));
for (int v = 0; v < NUM_SYNTH_VOICES; v++) {
SynthVoice *voice = &audio->synthVoices[v];
if (voice->volume == 0 || voice->frequency == 0) {
continue; // skip silent or inactive voices
}
// Compute source center X
float sourceCx = voice->sourceRect.x + voice->sourceRect.w * 0.5f;
if ((voice->volume == 0 && voice->smoothedAmp < 0.001f) || voice->frequency == 0)
continue;
float sourceCx = voice->sourceRect.x + TILE_SIZE * 0.5f;
float dx = sourceCx - listenerCx;
// Normalize for pan. If |dx| >= maxPanDistance → full left or full right.
float pan = dx / audio->maxPanDistance;
if (pan < -1.0f) pan = -1.0f;
if (pan > +1.0f) pan = +1.0f;
float pan = fmaxf(-1.0f, fminf(+1.0f, dx / audio->maxPanDistance));
float gainL, gainR;
compute_stereo_gains(pan, &gainL, &gainR);
gainL *= 0.7071f;
gainR *= 0.7071f;
// Optional: You could also attenuate overall volume with distance
// float dist = fabsf(dx);
// float distanceAtten = 1.0f - fminf(dist / audio->maxPanDistance, 1.0f);
// float finalVolume = (voice->volume / 255.0f) * distanceAtten;
// But for now, we’ll just use voice->volume for amplitude.
float dist = fabsf(dx);
float distanceAtten = 1.0f - fminf(dist / audio->maxPanDistance, 1.0f);
float targetAmp = (voice->volume / 255.0f) * distanceAtten;
float amp = (voice->volume / 255.0f);
double phaseInc = ((double) voice->frequency * 256.0) / (double) SAMPLE_RATE;
// Phase increment per sample‐frame:
// (freq * 256) / SAMPLE_RATE tells how many phase steps per mono-sample.
// Because we’re writing stereo, we still advance phase once per frame.
uint8_t phaseInc = (uint8_t)((voice->frequency * 256) / SAMPLE_RATE);
// Mix into each frame
for (int i = 0; i < frames; i++) {
float t = (float) voice->phase / 255.0f * 2.0f - 1.0f;
voice->smoothedAmp += (targetAmp - voice->smoothedAmp) * SMOOTHING_FACTOR;
float amp = voice->smoothedAmp;
double norm = voice->phase / 256.0;
double t = norm * 2.0 - 1.0;
float sample;
switch (voice->waveform) {
default:
case WAVE_SINE:
sample = sinf(voice->phase * 2.0f * M_PI / 256.0f);
break;
case WAVE_SQUARE:
sample = (t >= 0.0f) ? 1.0f : -1.0f;
sample = (t >= 0.0) ? 1.0f : -1.0f;
break;
case WAVE_SAWTOOTH:
sample = t;
sample = (float) t;
break;
case WAVE_TRIANGLE:
sample = (t < 0.0f) ? -t : t;
sample = (float) ((t < 0.0) ? -t : t);
break;
case WAVE_NOISE:
sample = ((float) rand() / RAND_MAX) * 2.0f - 1.0f;
sample = ((float) rand() / (float) RAND_MAX) * 2.0f - 1.0f;
break;
default:
sample = (float) sin(norm * 2.0 * M_PI);
break;
}
voice->phase += phaseInc;
if (voice->phase >= 256.0) voice->phase -= 256.0;
else if (voice->phase < 0.0) voice->phase += 256.0;
// Interleaved index: left = 2*i, right = 2*i + 1
int idxL = 2 * i;
int idxR = 2 * i + 1;
// Accumulate into buffer
outBuf[idxL] += sample * amp * gainL;
outBuf[idxR] += sample * amp * gainR;
voiceCounts[i]++;
}
}
for (int i = 0; i < frames; ++i) {
int count = voiceCounts[i];
if (count > 0) {
outBuf[2 * i + 0] /= count;
outBuf[2 * i + 1] /= count;
}
}
free(voiceCounts);
}
static uint32_t read_be_uint32(const uint8_t *data) {
return (data[0]<<24) | (data[1]<<16) | (data[2]<<8) | data[3];
}
static uint16_t read_be_uint16(const uint8_t *data) {
return (data[0]<<8) | data[1];
}
static uint32_t read_vlq(const uint8_t **ptr) {
uint32_t value = 0;
const uint8_t *p = *ptr;
while (*p & 0x80) {
value = (value << 7) | (*p++ & 0x7F);
}
value = (value << 7) | (*p++ & 0x7F);
*ptr = p;
return value;
}
void load_midi_file(const char *path) {
FILE *f = fopen(path, "rb");
if (!f) return;
fseek(f, 0, SEEK_END);
long size = ftell(f);
rewind(f);
uint8_t *data = malloc(size);
fread(data, 1, size, f);
fclose(f);
const uint8_t *ptr = data;
if (memcmp(ptr, "MThd", 4) != 0) return;
ptr += 8; // skip header length
uint16_t format = read_be_uint16(ptr); ptr += 2;
uint16_t nTracks = read_be_uint16(ptr); ptr += 2;
uint16_t ppqn = read_be_uint16(ptr); ptr += 2;
if (format != 0 || nTracks != 1) {
printf("Only Type 0 MIDI supported\n");
free(data);
return;
}
if (memcmp(ptr, "MTrk", 4) != 0) return;
uint32_t trackLen = read_be_uint32(ptr+4);
ptr += 8;
const uint8_t *trackEnd = ptr + trackLen;
float curTime = 0.0f;
uint32_t tempo = 500000; // default: 120 BPM
uint8_t lastStatus = 0;
while (ptr < trackEnd && midiEventCount < MAX_MIDI_EVENTS) {
uint32_t delta = read_vlq(&ptr);
curTime += (delta * (tempo / 1000000.0f)) / ppqn;
uint8_t status = *ptr;
if (status < 0x80) status = lastStatus;
else ptr++;
lastStatus = status;
if (status == 0xFF) {
uint8_t metaType = *ptr++;
uint32_t len = read_vlq(&ptr);
if (metaType == 0x51 && len == 3) {
tempo = (ptr[0]<<16 | ptr[1]<<8 | ptr[2]);
}
ptr += len;
} else if ((status & 0xF0) == 0x90 || (status & 0xF0) == 0x80) {
uint8_t note = *ptr++;
uint8_t vel = *ptr++;
midiEvents[midiEventCount++] = (MidiEvent){
.timeSec = curTime,
.type = (status & 0xF0) == 0x90 ? 0 : 1,
.note = note,
.velocity = vel
};
} else {
ptr += 2; // skip unknown
}
}
// Note: We did not normalize by active voices here, because each voice already
// uses its own volume. If you still want an automatic “divide by N active voices”,
// you would need to track active voices per‐frame, which is relatively expensive.
// In practice, you manage the volume per voice so clipping doesn’t occur.
}
free(data);
}