#include <string.h> // For memset, memcpy
#include <SDL2/SDL.h>
+#include "ay8910.h"
#include "log.h"
// Useful defines
//#define DEBUG
-#define SAMPLE_RATE (48000.0)
#define SAMPLES_PER_FRAME (SAMPLE_RATE / 60.0)
#define CYCLES_PER_SAMPLE (1024000.0 / SAMPLE_RATE)
// 32K ought to be enough for anybody
static SDL_AudioDeviceID device;
static bool soundInitialized = false;
static bool speakerState = false;
-static int16_t soundBuffer[SOUND_BUFFER_SIZE];
+static uint16_t soundBuffer[SOUND_BUFFER_SIZE];
static uint32_t soundBufferPos;
-static uint64_t lastToggleCycles;
-static SDL_cond * conditional = NULL;
-static SDL_mutex * mutex = NULL;
-static SDL_mutex * mutex2 = NULL;
-static int16_t sample;
+//static uint64_t lastToggleCycles;
+//static SDL_cond * conditional = NULL;
+//static SDL_mutex * mutex = NULL;
+//static SDL_mutex * mutex2 = NULL;
+static uint16_t sample;
static uint8_t ampPtr = 12; // Start with -2047 - +2047
static int16_t amplitude[17] = { 0, 1, 2, 3, 7, 15, 31, 63, 127, 255,
511, 1023, 2047, 4095, 8191, 16383, 32767 };
void SoundInit(void)
{
SDL_zero(desired);
- desired.freq = SAMPLE_RATE; // SDL will do conversion on the fly, if it can't get the exact rate. Nice!
- desired.format = AUDIO_S16SYS; // This uses the native endian (for portability)...
+ desired.freq = SAMPLE_RATE; // SDL will do conversion on the fly, if it can't get the exact rate. Nice!
+ desired.format = AUDIO_U16SYS; // This uses the native endian (for portability)...
desired.channels = 1;
- desired.samples = 512; // Let's try a 1/2K buffer (can always go lower)
+ desired.samples = 512; // Let's try a 1/2K buffer
desired.callback = SDLSoundCallback;
device = SDL_OpenAudioDevice(NULL, 0, &desired, &obtained, 0);
return;
}
- conditional = SDL_CreateCond();
- mutex = SDL_CreateMutex();
- mutex2 = SDL_CreateMutex();// Let's try real signalling...
+// conditional = SDL_CreateCond();
+// mutex = SDL_CreateMutex();
+// mutex2 = SDL_CreateMutex();// Let's try real signalling...
soundBufferPos = 0;
- lastToggleCycles = 0;
+// lastToggleCycles = 0;
sample = desired.silence; // ? wilwok ? yes
SDL_PauseAudioDevice(device, 0); // Start playback!
{
SDL_PauseAudioDevice(device, 1);
SDL_CloseAudioDevice(device);
- SDL_DestroyCond(conditional);
- SDL_DestroyMutex(mutex);
- SDL_DestroyMutex(mutex2);
+// SDL_DestroyCond(conditional);
+// SDL_DestroyMutex(mutex);
+// SDL_DestroyMutex(mutex2);
WriteLog("Sound: Done.\n");
}
}
//
// Sound card callback handler
//
+static uint32_t sndFrmCnt = 0;
+static uint32_t lastStarve = 0;
static void SDLSoundCallback(void * /*userdata*/, Uint8 * buffer8, int length8)
{
+sndFrmCnt++;
//WriteLog("SDLSoundCallback(): begin (soundBufferPos=%i)\n", soundBufferPos);
- // The sound buffer should only starve when starting which will cause it to
- // lag behind the emulation at most by around 1 frame...
- // (Actually, this should never happen since we fill the buffer beforehand.)
- // (But, then again, if the sound hasn't been toggled for a while, then this
- // makes perfect sense as the buffer won't have been filled at all!)
- // (Should NOT starve now, now that we properly handle frame edges...)
// Let's try using a mutex for shared resource consumption...
//Actually, I think Lock/UnlockAudio() does this already...
//WriteLog("SDLSoundCallback: soundBufferPos = %i\n", soundBufferPos);
- SDL_mutexP(mutex2);
+// SDL_mutexP(mutex2);
// Recast this as a 16-bit type...
- int16_t * buffer = (int16_t *)buffer8;
+ uint16_t * buffer = (uint16_t *)buffer8;
uint32_t length = (uint32_t)length8 / 2;
//WriteLog("SDLSoundCallback(): filling buffer...\n");
if (soundBufferPos < length)
{
+WriteLog("*** Sound buffer starved (%d short) *** [%d delta %d]\n", length - soundBufferPos, sndFrmCnt, sndFrmCnt - lastStarve);
+lastStarve = sndFrmCnt;
+#if 1
+ for(uint32_t i=0; i<length; i++)
+ buffer[i] = desired.silence;
+#else
// The sound buffer is starved...
for(uint32_t i=0; i<soundBufferPos; i++)
buffer[i] = soundBuffer[i];
// Reset soundBufferPos to start of buffer...
soundBufferPos = 0;
+#endif
}
else
{
// Free the mutex...
//WriteLog("SDLSoundCallback(): SDL_mutexV(mutex2)\n");
- SDL_mutexV(mutex2);
+// SDL_mutexV(mutex2);
// Wake up any threads waiting for the buffer to drain...
- SDL_CondSignal(conditional);
+// SDL_CondSignal(conditional);
//WriteLog("SDLSoundCallback(): end\n");
}
// This is called by the main CPU thread every ~21.666 cycles.
void WriteSampleToBuffer(void)
{
+#ifdef USE_NEW_AY8910
+ uint16_t s1 = AYGetSample(0);
+ uint16_t s2 = AYGetSample(1);
+ uint16_t adjustedMockingboard = s1 + s2;
+#else
+ int16_t s1, s2, s3, s4, s5, s6;
+ int16_t * bufPtrs[6] = { &s1, &s2, &s3, &s4, &s5, &s6 };
+ AY8910Update(0, bufPtrs, 1);
+ AY8910Update(1, &bufPtrs[3], 1);
+ int16_t adjustedMockingboard = (s1 / 8) + (s2 / 8) + (s3 / 8)
+ + (s4 / 8) + (s5 / 8) + (s6 / 8);
+#endif
+//need to do this *before* mixing, as by this time, it's too late and the sample is probably already oversaturated
+// adjustedMockingboard /= 8;
+
//WriteLog("WriteSampleToBuffer(): SDL_mutexP(mutex2)\n");
- SDL_mutexP(mutex2);
+// SDL_mutexP(mutex2);
// This should almost never happen, but, if it does...
while (soundBufferPos >= (SOUND_BUFFER_SIZE - 1))
{
//WriteLog("WriteSampleToBuffer(): Waiting for sound thread. soundBufferPos=%i, SOUNDBUFFERSIZE-1=%i\n", soundBufferPos, SOUND_BUFFER_SIZE-1);
- SDL_mutexV(mutex2); // Release it so sound thread can get it,
- SDL_mutexP(mutex); // Must lock the mutex for the cond to work properly...
- SDL_CondWait(conditional, mutex); // Sleep/wait for the sound thread
- SDL_mutexV(mutex); // Must unlock the mutex for the cond to work properly...
- SDL_mutexP(mutex2); // Re-lock it until we're done with it...
+// SDL_mutexV(mutex2); // Release it so sound thread can get it,
+// SDL_mutexP(mutex); // Must lock the mutex for the cond to work properly...
+// SDL_CondWait(conditional, mutex); // Sleep/wait for the sound thread
+// SDL_mutexV(mutex); // Must unlock the mutex for the cond to work properly...
+// SDL_mutexP(mutex2); // Re-lock it until we're done with it...
+ SDL_Delay(1);
}
- soundBuffer[soundBufferPos++] = sample;
+ SDL_LockAudioDevice(device);
+ soundBuffer[soundBufferPos++] = sample + adjustedMockingboard;
+ SDL_UnlockAudioDevice(device);
+
+// soundBuffer[soundBufferPos++] = sample;
//WriteLog("WriteSampleToBuffer(): SDL_mutexV(mutex2)\n");
- SDL_mutexV(mutex2);
+// SDL_mutexV(mutex2);
}
return;
speakerState = !speakerState;
- sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]);
+ sample = (speakerState ? amplitude[ampPtr] : 0);//-amplitude[ampPtr]);
}
projects / apple2 / blobdiff