ampler/audio.c

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// audio.c
void sound_src_frame(Sound_src *s, s8 *mix[]) {
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if s -> state == LOOPING do
for int i = 0; i < FRAME_SAMPLES; i += 1 do {
for int c = 0; c < CHANNELS; c += 1 do
mix[c][i] += s -> tracks[c][(int) s -> pos];
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s -> pos += s -> speed;
while s -> pos < 0.0f do
s -> pos += s -> len;
while s -> pos >= s -> len do
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s -> pos -= s -> len;
}
}
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int audio_frame(Ampler_state *state) {
int queued = SDL_GetQueuedAudioSize(state -> playdev);
queued /= sizeof s16; // queued is in bytes
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if queued > FRAME_SAMPLES * CHANNELS * 2 do return 0;
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// else puts("queued audio.");
// TODO: We should use CHANNELS here
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static s8 mix_l[FRAME_SAMPLES] = { 0 }, mix_r[FRAME_SAMPLES] = { 0 };
SDL_memset(mix_l, 0, sizeof mix_l);
SDL_memset(mix_r, 0, sizeof mix_r);
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s8 *mix[] = { mix_l, mix_r };
for int s = 0; s < arraylen(state -> sounds); s += 1 do
sound_src_frame(&(state -> sounds[s]), mix);
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const int vol = 100;
static s16 frame[FRAME_SAMPLES * CHANNELS] = { 0 };
for int i = 0; i < FRAME_SAMPLES; i += 1 do
for int t = 0; t < CHANNELS; t += 1 do
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frame[i * CHANNELS + t] = ((s16) (mix[t][i])) * vol;
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if SDL_QueueAudio(state -> playdev, frame, sizeof frame) do
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puts(SDL_GetError());
return 1; // audio was qued
}
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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 length = (bytes / sizeof s16) / chans;
// TODO: Actually find free sound_src
Sound_src *snd = &(state -> sounds[0]);
snd -> len = length;
snd -> speed = 1.0f;
snd -> pos = 0.0f;
snd -> state = LOOPING; // TOOD: Should be STOPPED
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for int i = 0; i < CHANNELS; i += 1 do
snd -> tracks[i] = malloc(snd -> len);
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for int i = 0; i < CHANNELS; i += 1 do
if !snd -> tracks[i] do
puts("fffuuuck");
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for int i = 0; i < snd -> len; i += 1 do
for int t = 0; t < chans; t += 1 do
snd -> tracks[t][i] = (s8) (buffer[i * chans + t] >> 8);
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SDL_FreeWAV((void *) buffer);
printf("loaded sample.wav %f seconds.\n",
((f32) length) / ((f32) SAMPLE_RATE));
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}
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));
}
*/
}