visualiser now runns on defaut input device

ampler.c

@@ -29,11 +29,21 @@ Ampler_state *init() {
 	if !(state -> playdev) do
 		puts(SDL_GetError());
 
+	state -> recdev = SDL_OpenAudioDevice(
+		NULL, // default device
+		1, // is capture
+		&want,
+		&have,
+		0 // no changes allowed
+	);
+	if !(state -> playdev) do
+		puts(SDL_GetError());
+
 	// TODO: iterate a directory to load samples
-	//load_sample(state, "mix.wav");
+	//load_sample(state, "./mix.wav");
-	//load_sample(state, "loop2.wav");
+	//load_sample(state, "./loop2.wav");
-	//load_sample(state, "loop.wav");
+	//load_sample(state, "./loop.wav");
-	//load_sample(state, "chord.wav");
+	//load_sample(state, "./chord.wav");
 
 	puts("init.");
 	return state;
@@ -60,6 +70,7 @@ int ampler_main(SDL_Window *w, SDL_Renderer *r, Ampler_state **u_data) {
 	// put results in state for last frame's perf, so that we can draw it
 	SDL_SetWindowTitle(w, "ampler");
 	SDL_PauseAudioDevice(state -> playdev, 0); // unpause audio
+	SDL_PauseAudioDevice(state -> recdev, 0); // unpause audio
 	udp_init(state);
 	while 1 do {
 		SDL_Event e;
@@ -80,6 +91,7 @@ int ampler_main(SDL_Window *w, SDL_Renderer *r, Ampler_state **u_data) {
 
 		if !remove("reload-trigger") do {
 			SDL_PauseAudioDevice(state -> playdev, 1); // pause audio
+			SDL_PauseAudioDevice(state -> recdev, 1); // pause audio
 			udp_quit(state);
 			puts("removed reload-trigger, reloading...");
 			return 0;

ampler.h

@@ -63,10 +63,12 @@ typedef struct Ampler_state Ampler_state;
 struct Ampler_state {
 	u32 size; // in bytes, off this struct
 	SDL_AudioDeviceID playdev;
+	SDL_AudioDeviceID recdev;
 	Sound_src sounds[64];
 	s32 played_audio_last_frame;
 	Udp_msg messages[64];
 	s8 frame_mix[CHANNELS][FRAME_SAMPLES];
+	s8 frame_rec[CHANNELS][FRAME_SAMPLES];
 };
 
 

audio.c

@@ -60,7 +60,31 @@ void sound_src_frame(Sound_src *s, s8 *mix[]) {
 	}
 }
 
+void rec_frame(Ampler_state *state) {
+	// const int frame_bytes = FRAME_SAMPLES * CHANNELS * sizeof(s16);
+	static s16 buf[FRAME_SAMPLES * CHANNELS * sizeof(s16)];
+
+	u32 samps = SDL_GetQueuedAudioSize(state->recdev) / sizeof(s16);
+
+	// TODO: this introduces a frame of lag, but should make sure we always
+	// deque a full frame, maybe find a less laggy way ?
+	if samps > FRAME_SAMPLES * CHANNELS * 2 do {
+		u32 bytes = SDL_DequeueAudio(state->recdev, buf, sizeof(buf));
+		// printf("%i %i %i samples\n", samps, bytes / sizeof(s16), sizeof(buf) / 2);
+	}
+
+	// TODO: test with stero recording input to make sure this works
+	for int i = 0; i < FRAME_SAMPLES; i += 1 do
+		for int c = 0; c < CHANNELS; c += 1 do {
+			state->frame_rec[c][i] = buf[i * CHANNELS + c] >> 8;
+		}
+}
+
 int audio_frame(Ampler_state *state) {
+	// TODO: it's unlikely but possible that we can miss recorded data because
+	// gur rec buffer was filled on a frame where we didn't play anything
+	rec_frame(state);
+
 	int queued = SDL_GetQueuedAudioSize(state -> playdev);
 	queued /= sizeof (s16); // queued is in bytes
 
@@ -107,7 +131,7 @@ void load_sample(Ampler_state *state, const char *file_name) {
 	s16 *buffer = NULL;
 	u32 bytes = 0;
 
-	SDL_LoadWAV(file_name, &spec, &buffer, &bytes);
+	SDL_LoadWAV(file_name, &spec, (u8 **) &buffer, &bytes);
 
 	if spec.format != AUDIO_S16 do {
 		puts("error: sample.wav is not s16");

draw.c

@@ -40,6 +40,10 @@ void draw_frame(SDL_Window *w, SDL_Renderer *r, Ampler_state *state) {
 	for int x = 0; x < FRAME_SAMPLES - 1; x += 1 do {
 		u8 cur = state->frame_mix[0][x] + 128;
 		u8 next = state->frame_mix[0][x + 1] + 128;
+
+		cur = state->frame_rec[0][x] + 128;
+		next = state->frame_rec[0][x + 1] + 128;
+
 		markov[cur][next] += 1;
 	}