switched to floats for speed+pos

ampler.c

@@ -65,7 +65,7 @@ int ampler_main(SDL_Window *w, SDL_Renderer *r, Ampler_state **u_data) {
 			if e.type == SDL_QUIT do
 				return 1;
 
-		audio_frame(state);
+		state -> played_audio_last_frame = audio_frame(state);
 		draw_frame(w, r, state);
 
 		SDL_Delay(1);

ampler.h

@@ -5,6 +5,8 @@
 #define for for(
 #define do )
 
+#define or ||
+
 typedef int8_t s8;
 typedef int16_t s16;
 typedef int32_t s32;
@@ -25,10 +27,21 @@ typedef double f64;
 #define CHANNELS (2)
 #define FRAME_SAMPLES (SAMPLE_RATE / 60)
 
+typedef struct Sound_src Sound_src;
+struct Sound_src {
+	s32 start; // in samples
+	s32 len; // in samples
+	f32 pos; // position in samples
+	f32 speed; // in samples, per sample, can be negative
+	enum { FREE = 0, STOPPED, PLAYING, LOOPING } state;
+};
+
 typedef struct Ampler_state Ampler_state;
 struct Ampler_state {
 	u32 size; // in bytes, off this struct
 	SDL_AudioDeviceID playdev;
 	s8 tracks[CHANNELS][SAMPLE_RATE * 480]; // 8 minutes
+	Sound_src sounds[64];
+	s32 played_audio_last_frame;
 };
 

audio.c

@@ -1,5 +1,19 @@
 // audio.c
 
+void sound_src_frame(Sound_src *s, s8 *tracks[], s8 *mix[]) {
+	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] += tracks[c][(int) (s -> start + s -> pos)];
+
+		s -> pos += s -> speed;
+		while s -> pos < 0.0f do
+			s -> pos += s -> len;
+		while s -> pos > s -> len do
+			s -> pos -= s -> len;
+	}
+}
+
 int audio_frame(Ampler_state *state) {
 	int queued = SDL_GetQueuedAudioSize(state -> playdev);
 	queued /= sizeof s16; // queued is in bytes
@@ -7,14 +21,20 @@ int audio_frame(Ampler_state *state) {
 	if queued > FRAME_SAMPLES * CHANNELS * 2 do return 0;
 	// else puts("queued audio.");
 
-	// TODO: ITERATE SOUNDS AND MIX IN SEP CHANNELS
+	static s8 mix_l[FRAME_SAMPLES] = { 0 }, mix_r[FRAME_SAMPLES] = { 0 };
-	// TODO: THEN, CONVERT TO INTERLEAVED AND OUTPUT
+	SDL_memset(mix_l, 0, sizeof mix_l);
+	SDL_memset(mix_r, 0, sizeof mix_r);
 
-	static s16 frame[SAMPLE_RATE * CHANNELS * 7] = { 0 };
+	s8 *mix[] = { mix_l, mix_r };
-	const int vol = 64;
+	s8 *tracks[] = { state -> tracks[0], state -> tracks[1] };
-	for int i = 0; i < arraylen(frame) / CHANNELS; i += 1 do
+	for int s = 0; s < arraylen(state -> sounds); s += 1 do
+		sound_src_frame(&(state -> sounds[s]), tracks, mix);
+
+	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
-			frame[i * CHANNELS + t] = ((s16) (state -> tracks[t][i])) * vol;
+			frame[i * CHANNELS + t] = ((s16) (mix[t][i])) * vol;
 
 	if SDL_QueueAudio(state -> playdev, frame, sizeof frame) do
 		puts(SDL_GetError());
@@ -46,6 +66,13 @@ void load_track(Ampler_state *state) {
 		for int t = 0; t < chans; t += 1 do
 			state -> tracks[t][i] = (s8) (buffer[i * chans + t] >> 8);
 
+	// TODO: Find actually free sound_src
+	state -> sounds[0].start = 0;
+	state -> sounds[0].len = len;
+	state -> sounds[0].speed = 1.0f;
+	state -> sounds[0].pos = 0.0f;
+	state -> sounds[0].state = LOOPING;
+
 	SDL_FreeWAV((void *) buffer);
 }
 

draw.c

@@ -10,8 +10,14 @@ void draw_frame(SDL_Window *w, SDL_Renderer *r, Ampler_state *state) {
 		for int x = 0; x < 512; x += 1 do {
 			const int y = 128 + 256 * t;
 			const int val = state -> tracks[t][x * space];
-			SDL_RenderDrawLine(r, x, y + val / 8, x, y + val);
+			SDL_RenderDrawLine(r, y + val / 8, x, y + val, x);
-			//SDL_RenderDrawPoint(r, x, y + val);
+			//SDL_RenderDrawPoint(r, y + val, x);
+		}
+
+	if state -> played_audio_last_frame do
+	for int i = 0; i < arraylen(state -> sounds); i += 1 do {
+		const int x = (state -> sounds[i].start + state -> sounds[i].pos) / space;
+		SDL_RenderDrawLine(r, 0, x, 512, x);
 	}
 
 	SDL_RenderPresent(r);