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ampler
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visualiser now runns on defaut input device

master
bx 2021-12-27 06:23:29 +00:00
parent 6f01e60b10
commit 6b3c8210b5
4 changed files with 47 additions and 5 deletions
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20
ampler.c
View File

@ -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, "loop2.wav");
//load_sample(state, "loop.wav");
//load_sample(state, "chord.wav");
//load_sample(state, "./mix.wav");
//load_sample(state, "./loop2.wav");
//load_sample(state, "./loop.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;

2
ampler.h
View File

@ -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];
};

26
audio.c
View File

@ -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");

4
draw.c
View File

@ -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;
}