sounds now have start and end fields
parent
2680732685
commit
0bb207bbca
3
ampler.c
3
ampler.c
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@ -67,7 +67,8 @@ int ampler_main(SDL_Window *w, SDL_Renderer *r, Ampler_state **u_data) {
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state -> played_audio_last_frame = audio_frame(state);
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state -> played_audio_last_frame = audio_frame(state);
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draw_frame(w, r, state);
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draw_frame(w, r, state);
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// CLEAR UDP MESSAGES
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// CLEAR UDP MESSAGES, but only after we've played them
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if state -> played_audio_last_frame do
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foreach_ptr(Udp_msg, msg, state -> messages)
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foreach_ptr(Udp_msg, msg, state -> messages)
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msg -> state = MSG_FREE;
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msg -> state = MSG_FREE;
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4
ampler.h
4
ampler.h
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@ -40,9 +40,11 @@ for ; N##_i < arraylen(ARRAY); N##_i += 1, N = &((ARRAY)[N##_i]) do
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typedef struct Sound_src Sound_src;
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typedef struct Sound_src Sound_src;
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struct Sound_src {
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struct Sound_src {
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s8 *tracks[CHANNELS];
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s8 *tracks[CHANNELS];
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s32 len; // in samples
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s32 track_len; // size of track buffers, in samples
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f32 pos; // position in samples
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f32 pos; // position in samples
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f32 speed; // in samples, per sample, can be negative
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f32 speed; // in samples, per sample, can be negative
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s32 start; // play start in samples
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s32 end; // play end in samples
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enum { SND_FREE = 0, SND_STOPPED, SND_PLAYING, SND_LOOPING, } state;
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enum { SND_FREE = 0, SND_STOPPED, SND_PLAYING, SND_LOOPING, } state;
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};
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};
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56
audio.c
56
audio.c
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@ -3,17 +3,48 @@
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// TODO: add something to make sure, we KNOW if we've dropped audio
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// TODO: add something to make sure, we KNOW if we've dropped audio
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// eg. buffer has gone unfilled for a frame or more
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// eg. buffer has gone unfilled for a frame or more
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void trigger_sounds(Ampler_state *state) {
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// TODO: Check sound names
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// TODO: func to decode orca numbers
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// TODO: NOTE SPEED
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foreach_ptr(Udp_msg, m, state -> messages)
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if m -> state == MSG_TRIGGER do
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foreach_ptr(Sound_src, s, state -> sounds)
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if s -> state == SND_STOPPED do {
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s -> state = SND_PLAYING;
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s -> speed = 7.01f;
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s -> pos = 0.0f;
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break;
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}
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else if s -> state == SND_PLAYING do {
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s -> speed = -1.51f;
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break;
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}
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}
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void sound_src_frame(Sound_src *s, s8 *mix[]) {
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void sound_src_frame(Sound_src *s, s8 *mix[]) {
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if s -> state == SND_LOOPING do
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if s -> state == SND_LOOPING or s -> state == SND_PLAYING do
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for int i = 0; i < FRAME_SAMPLES; i += 1 do {
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for int i = 0; i < FRAME_SAMPLES; i += 1 do {
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for int c = 0; c < CHANNELS; c += 1 do
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for int c = 0; c < CHANNELS; c += 1 do
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mix[c][i] += s -> tracks[c][(int) s -> pos];
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mix[c][i] += s -> tracks[c][(int) s -> pos];
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s -> pos += s -> speed;
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s -> pos += s -> speed;
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while s -> pos < 0.0f do
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s -> pos += s -> len;
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if s -> state == SND_LOOPING do {
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while s -> pos >= s -> len do
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while s -> pos < s -> start do
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s -> pos -= s -> len;
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s -> pos += (s->end - s->start);
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while s -> pos >= s -> end do
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s -> pos -= (s->end - s->start);
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}
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// TODO: THIS WILL BREAK REVERSE SOUNDS
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// FIXME: CHECK SIGN OF SPEED
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if s -> state == SND_PLAYING do
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if s->pos < s->start or s->pos >= s->end do {
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s -> pos = s->start;
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s -> state = SND_STOPPED;
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}
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}
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}
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}
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}
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@ -24,6 +55,8 @@ int audio_frame(Ampler_state *state) {
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if queued > FRAME_SAMPLES * CHANNELS * 2 do return 0;
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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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// else puts("queued audio.");
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trigger_sounds(state);
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// TODO: We should use CHANNELS here
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// TODO: We should use CHANNELS here
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static s8 mix_l[FRAME_SAMPLES] = { 0 }, mix_r[FRAME_SAMPLES] = { 0 };
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static s8 mix_l[FRAME_SAMPLES] = { 0 }, mix_r[FRAME_SAMPLES] = { 0 };
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SDL_memset(mix_l, 0, sizeof mix_l);
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SDL_memset(mix_l, 0, sizeof mix_l);
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@ -71,20 +104,25 @@ void load_track(Ampler_state *state) {
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const int chans = spec.channels;
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const int chans = spec.channels;
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const int length = (bytes / sizeof s16) / chans;
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const int length = (bytes / sizeof s16) / chans;
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snd -> len = length;
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snd -> track_len = length;
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snd -> speed = 1.0f;
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snd -> speed = 1.0f;
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snd -> pos = 0.0f;
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snd -> pos = 0.0f;
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// snd -> state = SND_LOOPING;
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snd -> start = 0;
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snd -> end = snd->track_len;
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snd -> state = SND_STOPPED;
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snd -> state = SND_STOPPED;
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snd -> state = SND_LOOPING;
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snd -> end /= 2;
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snd -> start = snd -> end / 2;
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for int i = 0; i < CHANNELS; i += 1 do
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for int i = 0; i < CHANNELS; i += 1 do
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snd -> tracks[i] = malloc(snd -> len);
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snd -> tracks[i] = malloc(snd -> track_len);
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for int i = 0; i < CHANNELS; i += 1 do
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for int i = 0; i < CHANNELS; i += 1 do
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if !(snd -> tracks[i]) do
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if !(snd -> tracks[i]) do
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puts("fffuuuck");
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puts("fffuuuck");
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for int i = 0; i < snd -> len; i += 1 do
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for int i = 0; i < snd -> track_len; i += 1 do
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for int t = 0; t < chans; t += 1 do
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for int t = 0; t < chans; t += 1 do
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snd -> tracks[t][i] = (s8) (buffer[i * chans + t] >> 8);
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snd -> tracks[t][i] = (s8) (buffer[i * chans + t] >> 8);
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6
draw.c
6
draw.c
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@ -5,6 +5,7 @@ void draw_frame(SDL_Window *w, SDL_Renderer *r, Ampler_state *state) {
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//SDL_SetRenderDrawColor(r, 10, 10, 10, 100);
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//SDL_SetRenderDrawColor(r, 10, 10, 10, 100);
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SDL_RenderClear(r);
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SDL_RenderClear(r);
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// TODO: DRAW WITH X SCALE FACTOR
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// TODO: DRAW SMOLER
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// TODO: DRAW SMOLER
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SDL_SetRenderDrawColor(r, 0xFF, 0x00, 0x4D, 255);
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SDL_SetRenderDrawColor(r, 0xFF, 0x00, 0x4D, 255);
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const space = (44100 * 8) / 512;
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const space = (44100 * 8) / 512;
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@ -20,9 +21,12 @@ void draw_frame(SDL_Window *w, SDL_Renderer *r, Ampler_state *state) {
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}
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}
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if state -> played_audio_last_frame do
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if state -> played_audio_last_frame do
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foreach_ptr(Sound_src, snd, state -> sounds) {
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foreach_ptr(Sound_src, snd, state -> sounds)
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if snd->state == SND_PLAYING or snd->state == SND_LOOPING do {
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const int x = snd -> pos / space;
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const int x = snd -> pos / space;
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SDL_RenderDrawLine(r, x, 0, x, 512);
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SDL_RenderDrawLine(r, x, 0, x, 512);
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SDL_RenderDrawLine(r, snd->start / space, 0, snd->start / space, 512);
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SDL_RenderDrawLine(r, snd->end / space, 0, snd->end / space, 512);
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}
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}
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SDL_RenderPresent(r);
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SDL_RenderPresent(r);
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