ampler/audio.c

70 lines
1.7 KiB
C

// audio.c
int audio_frame(Ampler_state *state) {
int queued = SDL_GetQueuedAudioSize(state -> playdev);
queued /= sizeof s16; // queued is in bytes
if queued > FRAME_SAMPLES * CHANNELS * 2 do return 0;
// else puts("queued audio.");
// TODO: ITERATE SOUNDS AND MIX IN SEP CHANNELS
// TODO: THEN, CONVERT TO INTERLEAVED AND OUTPUT
static s16 frame[SAMPLE_RATE * CHANNELS * 7] = { 0 };
const int vol = 64;
for int i = 0; i < arraylen(frame) / CHANNELS; i += 1 do
for int t = 0; t < CHANNELS; t += 1 do
frame[i * CHANNELS + t] = ((s16) (state -> tracks[t][i])) * vol;
if SDL_QueueAudio(state -> playdev, frame, sizeof frame) do
puts(SDL_GetError());
return 1; // audio was qued
}
void load_track(Ampler_state *state) {
SDL_AudioSpec spec;
s16 *buffer = NULL;
u32 bytes = 0;
SDL_LoadWAV("sample.wav", &spec, &buffer, &bytes);
if spec.format != AUDIO_S16 do {
puts("error: sample.wav is not s16");
return;
}
const int chans = spec.channels;
const int len = (bytes / sizeof s16) / chans;
// TODO: VALIDATE LENGTH + TRACKS
printf("loading sample.wav %f seconds.\n",
((f32) len) / ((f32) SAMPLE_RATE));
for int i = 0; i < len; i += 1 do
for int t = 0; t < chans; t += 1 do
state -> tracks[t][i] = (s8) (buffer[i * chans + t] >> 8);
SDL_FreeWAV((void *) buffer);
}
void load_track_f32() { // idk if we're ever gonna be loading floats
/*
f32 max = 0.0f;
for int i = 0; i < len; i++ do
if buffer[i] > max do
max = buffer[i];
else if buffer[i] < -max do
max = -buffer[i];
printf("%f max\n", max);
for int i = 0; i < len; i += 2 do {
buffer[i] /= max;
buffer[i + 1] /= max;
state -> track_l[i] = ((s8)(buffer[i] * 127.0f));
state -> track_r[i] = ((s8)(buffer[i + 1] * 127.0f));
}
*/
}