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