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More precise sound clock calculation

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This commit is contained in:
tudnai 2022-07-14 10:53:00 -07:00
parent 99c2dc2828
commit 4388474aed
2 changed files with 23 additions and 44 deletions
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6
src/cpu/6502.c
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@ -321,10 +321,10 @@ void m6502_Run() {
//#ifdef CLK_WAIT
// unsigned long long elpased = (unsigned long long)-1LL;
//#endif
m6502.clktime += m6502.clkfrm;
m6502.clkfrm = decrement(m6502.clkfrm, clk_6502_per_frm_max);
m6502.clkfrm = 0;
if( diskAccelerator_count ) {
if( --diskAccelerator_count <= 0 ) {
// make sure we only adjust clock once to get back to normal

61
src/dev/audio/speaker.c
View File

@ -33,7 +33,13 @@
#include "disk.h" // to be able to disable disk acceleration
#define SPKR_OVERSAMPLING 32 // 8 and 10 sounds ok, 16 is better, 32 is really good, 64 is not better than 32
// 8 and 10 sounds ok
// 16 is better
// 32 is really good
// 64 dow not sound better much than 32
// 128 Relally good!
// 256 sounds really amazing!
#define SPKR_OVERSAMPLING 128
#define min(x,y) (x) < (y) ? (x) : (y)
@ -94,7 +100,9 @@ int spkr_last_level = SPKR_LEVEL_ZERO;
//static const int ema_len_soft = 20;
//static const int ema_len_supersoft = 40;
#if (SPKR_OVERSAMPLING >= 64)
#if (SPKR_OVERSAMPLING >= 128)
int spkr_ema_len = 256;
#elif (SPKR_OVERSAMPLING >= 64)
int spkr_ema_len = 128;
#elif (SPKR_OVERSAMPLING >= 32)
int spkr_ema_len = 64;
@ -108,7 +116,7 @@ int spkr_ema_len = 20;
int spkr_ema_len = 16;
#endif
#define BUFFER_COUNT 16
#define BUFFER_COUNT 32
#define SPKR_CHANNELS 2
ALuint spkr_src [SOURCES_COUNT] = { 0, 0, 0, 0 };
@ -245,12 +253,10 @@ FILE * spkr_debug_ema_file = NULL;
#endif
void spkr_fade(float fadeLevel) {
void spkr_fade(float fadeLevel, int idx) {
// Fade
int idx = 0;
float fadeSlope = fadeLevel / spkr_buf_size * SPKR_CHANNELS;
float fadeSlope = fadeLevel / (spkr_buf_size - idx) * SPKR_CHANNELS;
// while ( ( fabs(fadeLevel) >= fabs(fadeSlope) ) && ( idx < SPKR_BUF_SLOT_SIZE(1) ) ) {
while ( idx < SPKR_BUF_SLOT_SIZE(1) ) {
@ -264,7 +270,7 @@ void spkr_fade(float fadeLevel) {
// // Fill with Zero to avoid pops
// memset(spkr_samples + idx, 0, SPKR_BUF_SLOT_SIZE(1));
spkr_level = fadeLevel;
spkr_level = SPKR_LEVEL_ZERO;
}
@ -578,6 +584,7 @@ void spkr_toggle() {
// spkr_sample_idx = round( (spkr_clk + m6502.clkfrm) / ( MHZ(default_MHz_6502) / (double)spkr_sample_rate)) * SPKR_CHANNELS;
spkr_sample_idx = round( (double)(spkr_clk + m6502.clkfrm) * multiplier ) * SPKR_CHANNELS;
spkr_sample_idx %= SPKR_BUF_SLOT_SIZE(BUFFER_COUNT);
// if play stopped, we should make sure we are not creating a false initial quare wave
@ -587,12 +594,6 @@ void spkr_toggle() {
spkr_play_time = spkr_play_timeout;
unsigned spkr_sample_idx_diff = spkr_sample_idx - spkr_sample_last_idx;
if ( (int)spkr_sample_idx_diff < 0 ) {
spkr_sample_idx_diff = UINT_MAX - spkr_sample_idx_diff;
}
if ( spkr_state ) {
// down edge
spkr_state = 0;
@ -608,10 +609,6 @@ void spkr_toggle() {
spkr_toggle_square(SPKR_LEVEL_MAX);
}
//spkr_samples[sample_idx] = spkr_level;
for ( int i = spkr_sample_idx; i < spkr_buf_size + spkr_extra_buf; i++ ) {
spkr_samples[i] = spkr_level;
}
}
}
@ -923,7 +920,7 @@ void spkr_update() {
// Fix: Unqueue was not working properly some cases, so we need to monitor
// queued elements and if there are too many, let's just wait
#define SPKR_MAX_QUEUED 10
#define SPKR_MAX_QUEUED 25
ALint queued = 0;
alGetSourcei ( spkr_src[SPKR_SRC_GAME_SFX], AL_BUFFERS_QUEUED, &queued );
al_check_error();
@ -950,7 +947,7 @@ void spkr_update() {
// Need to Cool Down Speaker -- Soft Fade to Zero
else {
spkr_fade(spkr_level);
spkr_fade(spkr_level, 0);
spkr_filter();
#ifdef SPKR_DEBUG
@ -980,7 +977,7 @@ void spkr_update() {
// Normal Sound Buffer Feed
else {
// finish the aquare wave
// finish square wave
spkr_finish_square(spkr_buf_size);
// digital filtering the audio stream -- most notably smoothing
spkr_filter();
@ -999,27 +996,9 @@ void spkr_update() {
memcpy(spkr_samples, spkr_samples + SPKR_BUF_SLOT_SIZE(1), SPKR_BUF_SLOT_SIZE(1) );
memset(spkr_samples + SPKR_BUF_SLOT_SIZE(1), 0, SPKR_BUF_SLOT_SIZE(1) );
spkr_sample_idx -= spkr_buf_size;
while (spkr_sample_idx >= spkr_buf_size) {
spkr_sample_idx -= spkr_buf_size;
}
if ((int)spkr_sample_idx < 0) {
spkr_sample_idx = 0;
}
spkr_sample_idx = 0;
spkr_sample_last_idx = 0;
spkr_sample_last_idx -= spkr_buf_size;
while (spkr_sample_last_idx >= spkr_buf_size) {
spkr_sample_last_idx -= spkr_buf_size;
}
if ((int)spkr_sample_last_idx < 0) {
spkr_sample_last_idx = 0;
}
if (spkr_sample_last_idx >= spkr_buf_size) {
printf("spkr_sample_last_idx >= spkr_buf_size\n");
}
// make sure it never goes below 0 (never overflows)
if ( (int)spkr_play_time < 0 ) {
spkr_play_time = 0;