Implemented stereo sound.
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b44d85a299
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@ -43,11 +43,9 @@ BRK
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~Time.second ~current.second NEQ #01 JNZ BRK
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~Time.second =current.second
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~Time.second #03 AND ^$no-sound JNZ
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#ffff =Audio.ch1adsr
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#1444 =Audio.ch1adsr
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~Time.second #1d ADD =Audio.ch1pitch
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#ff =Audio.ch1vol
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$no-sound
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~Time.second #03 AND #4b MUL #0f ADD =Audio.ch1vol
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( clear )
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#0080 SCALEX #0080 SCALEY ~needles.sx ~needles.sy #00 ,draw-line JSR2
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@ -51,7 +51,8 @@ Uint8 font[][8] = {
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#define SAMPLE_FREQUENCY 48000
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static Uint32 note_periods[12] = { /* middle C (C4) is note 60 */
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static Uint32 note_periods[12] = {
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/* middle C (C4) is note 60 */
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(Uint32)0xfa7e * SAMPLE_FREQUENCY, /* C-1 */
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(Uint32)0xec6f * SAMPLE_FREQUENCY,
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(Uint32)0xdf2a * SAMPLE_FREQUENCY, /* D-1 */
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@ -66,11 +67,13 @@ static Uint32 note_periods[12] = { /* middle C (C4) is note 60 */
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(Uint32)0x84b2 * SAMPLE_FREQUENCY /* B-1 */
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};
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static struct audio_channel {
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typedef struct audio_channel {
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Uint32 period, count;
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Sint32 age, a, d, s, r;
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Sint16 volume, value;
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} channels[4];
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Sint16 value[2];
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Sint8 volume[2], phase;
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} Channel;
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Channel channels[4];
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static SDL_Window *gWindow;
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static SDL_Renderer *gRenderer;
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@ -236,39 +239,58 @@ togglezoom(Uxn *u)
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redraw(pixels, u);
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}
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Sint16
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audio_envelope(Channel *c) {
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if (c->age < c->a)
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return 0x0888 * c->age / c->a;
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else if (c->age < c->d)
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return 0x0444 * (2 * c->d - c->a - c->age) / (c->d - c->a);
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else if (c->age < c->s)
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return 0x0444;
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else if (c->age < c->r)
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return 0x0444 * (c->r - c->age) / (c->r - c->s);
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else
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return 0x0000;
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}
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void
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audio_callback(void* userdata, Uint8* stream, int len) {
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Sint16 *samples = (Sint16 *) stream;
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int i, j;
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len >>= 1; /* use len for number of samples, not bytes */
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for (j = 0; j < len; ++j) samples[j] = 0;
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len >>= 2; /* use len for number of samples, not bytes */
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for (j = len * 2 - 1; j >= 0; --j) samples[j] = 0;
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for (i = 0; i < 4; ++i) {
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struct audio_channel *c = &channels[i];
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if (!c->volume) continue;
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Channel *c = &channels[i];
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if (c->period < (1 << 20)) continue;
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for (j = 0; j < len; ++j) {
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c->age += 1;
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c->count += 1 << 20;
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while (c->count > c->period) {
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int mul = c->value < 0 ? c->volume : -c->volume;
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Sint16 mul;
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c->count -= c->period;
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if (c->age < c->a)
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c->value = mul * c->age / c->a;
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else if (c->age < c->d)
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c->value = mul * (2 * c->d - c->a - c->age) / 2 / (c->d - c->a);
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else if (c->age < c->s)
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c->value = mul / 2;
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else if (c->age < c->r)
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c->value = mul * (c->r - c->age) / 2 / (c->r - c->s);
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else
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c->volume = c->value = 0;
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c->phase = !c->phase;
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mul = (c->phase * 2 - 1) * audio_envelope(c);
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c->value[0] = mul * c->volume[0];
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c->value[1] = mul * c->volume[1];
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}
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samples[j] += c->value;
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samples[j * 2] += c->value[0];
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samples[j * 2 + 1] += c->value[1];
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}
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}
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(void) userdata;
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}
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void
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silence(void) {
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int i;
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for (i = 0; i < 4; ++i) {
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Channel *c = &channels[i];
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c->volume[0] = 0;
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c->volume[1] = 0;
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c->period = 0;
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}
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}
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void
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quit(void)
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{
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@ -301,11 +323,12 @@ init(void)
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if(!(pixels = (Uint32 *)malloc(WIDTH * HEIGHT * sizeof(Uint32))))
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return error("Pixels", "Failed to allocate memory");
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clear(pixels);
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silence();
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SDL_StartTextInput();
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SDL_ShowCursor(SDL_DISABLE);
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as.freq = SAMPLE_FREQUENCY;
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as.format = AUDIO_S16;
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as.channels = 1;
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as.channels = 2;
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as.callback = audio_callback;
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as.samples = 2048;
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audio_id = SDL_OpenAudioDevice(NULL, 0, &as, NULL, 0);
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@ -474,17 +497,18 @@ audio_poke(Uxn *u, Uint16 ptr, Uint8 b0, Uint8 b1)
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{
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Uint8 *m = u->ram.dat;
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if (b0 & 1) {
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Uint16 channel_addr = ptr + (b0 & 0x6);
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struct audio_channel *c = &channels[(b0 & 0x6) >> 1];
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Uint16 addr = ptr + (b0 & 0x6);
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Channel *c = &channels[(b0 & 0x6) >> 1];
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SDL_LockAudioDevice(audio_id);
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c->period = note_periods[m[channel_addr + 8] % 12] >> (m[channel_addr + 8] / 12);
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c->period = note_periods[m[addr + 8] % 12] >> (m[addr + 8] / 12);
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c->count %= c->period;
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c->volume = m[channel_addr + 9] << 5;
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c->volume[0] = (m[addr + 9] >> 4) & 0xf;
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c->volume[1] = m[addr + 9] & 0xf;
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c->age = 0;
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c->a = ((m[channel_addr] >> 4) & 0xf) * (SAMPLE_FREQUENCY >> 4);
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c->d = c->a + (m[channel_addr] & 0xf) * (SAMPLE_FREQUENCY >> 4);
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c->s = c->d + ((m[channel_addr + 1] >> 4) & 0xf) * (SAMPLE_FREQUENCY >> 4);
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c->r = c->s + (m[channel_addr + 1] & 0xf) * (SAMPLE_FREQUENCY >> 4);
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c->a = (SAMPLE_FREQUENCY >> 4) * ((m[addr] >> 4) & 0xf);
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c->d = c->a + (SAMPLE_FREQUENCY >> 4) * (m[addr] & 0xf);
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c->s = c->d + (SAMPLE_FREQUENCY >> 4) * ((m[addr + 1] >> 4) & 0xf);
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c->r = c->s + (SAMPLE_FREQUENCY >> 4) * (m[addr + 1] & 0xf);
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SDL_UnlockAudioDevice(audio_id);
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}
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return b1;
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