// STILL TO DO:
//
// - Figure out why it's losing samples (Bard's Tale) [DONE]
+// - Figure out why it's playing too fast
//
#include "sound.h"
#include <SDL.h>
#include "log.h"
+// Useful defines
+
+//#define DEBUG
+//#define WRITE_OUT_WAVE
+
+// This is odd--seems to be working properly now! Maybe a bug in the SDL sound code?
+// Actually, it still doesn't sound right... Sounds too slow now. :-/
+// But then again, it's difficult to tell. Sometimes it slows waaaaaay down, but generally
+// seems to be OK other than that
+// Also, it could be that the discrepancy in pitch is due to the V65C02 and it's lack of
+// cycle accuracy...
//#define SAMPLE_RATE (44100.0)
#define SAMPLE_RATE (48000.0)
#define SAMPLES_PER_FRAME (SAMPLE_RATE / 60.0)
+// This works for AppleWin but not here... ??? WHY ???
+// ~ 21
#define CYCLES_PER_SAMPLE (1024000.0 / SAMPLE_RATE)
-#define SOUND_BUFFER_SIZE (8192)
-//#define AMPLITUDE (16) // -32 - +32 seems to be plenty loud!
+// ~ 17 (lower pitched than above...!)
+// Makes sense, as this is the divisor for # of cycles passed
+//#define CYCLES_PER_SAMPLE (800000.0 / SAMPLE_RATE)
+// This seems about right, compared to AppleWin--but AW runs @ 1.024 MHz
+// 23 (1.024) vs. 20 (0.900)
+//#define CYCLES_PER_SAMPLE (900000.0 / SAMPLE_RATE)
+//nope, too high #define CYCLES_PER_SAMPLE (960000.0 / SAMPLE_RATE)
+//#define CYCLES_PER_SAMPLE 21
+//#define SOUND_BUFFER_SIZE (8192)
+#define SOUND_BUFFER_SIZE (32768)
// Global variables
// Local variables
-static SDL_AudioSpec desired;
+static SDL_AudioSpec desired, obtained;
static bool soundInitialized = false;
static bool speakerState = false;
-static uint8 soundBuffer[SOUND_BUFFER_SIZE];
+static int16 soundBuffer[SOUND_BUFFER_SIZE];
static uint32 soundBufferPos;
-static uint32 sampleBase;
static uint64 lastToggleCycles;
-static uint64 samplePosition;
static SDL_cond * conditional = NULL;
static SDL_mutex * mutex = NULL;
static SDL_mutex * mutex2 = NULL;
-static uint8 ampPtr = 5;
-static uint16 amplitude[17] = { 0, 1, 2, 4, 8, 16, 32, 64, 128, 256, 512, 1024, 2048,
- 4096, 8192, 16384, 32768 };
+static int16 sample;
+static uint8 ampPtr = 14; // Start with -16 - +16
+static int16 amplitude[17] = { 0, 1, 2, 3, 7, 15, 31, 63, 127, 255, 511, 1023, 2047,
+ 4095, 8191, 16383, 32767 };
+#ifdef WRITE_OUT_WAVE
+static FILE * fp = NULL;
+#endif
// Private function prototypes
#endif
desired.freq = SAMPLE_RATE; // SDL will do conversion on the fly, if it can't get the exact rate. Nice!
- desired.format = AUDIO_S8; // This uses the native endian (for portability)...
-// desired.format = AUDIO_S16SYS; // This uses the native endian (for portability)...
+// desired.format = AUDIO_S8; // This uses the native endian (for portability)...
+ desired.format = AUDIO_S16SYS; // This uses the native endian (for portability)...
desired.channels = 1;
// desired.samples = 4096; // Let's try a 4K buffer (can always go lower)
// desired.samples = 2048; // Let's try a 2K buffer (can always go lower)
desired.samples = 1024; // Let's try a 1K buffer (can always go lower)
desired.callback = SDLSoundCallback;
- if (SDL_OpenAudio(&desired, NULL) < 0) // NULL means SDL guarantees what we want
+// if (SDL_OpenAudio(&desired, NULL) < 0) // NULL means SDL guarantees what we want
+//When doing it this way, we need to check to see if we got what we asked for...
+ if (SDL_OpenAudio(&desired, &obtained) < 0)
{
WriteLog("Sound: Failed to initialize SDL sound.\n");
return;
conditional = SDL_CreateCond();
mutex = SDL_CreateMutex();
mutex2 = SDL_CreateMutex();// Let's try real signalling...
- SDL_mutexP(mutex); // Must lock the mutex for the cond to work properly...
soundBufferPos = 0;
- sampleBase = 0;
lastToggleCycles = 0;
- samplePosition = 0;
+ sample = desired.silence; // ? wilwok ? yes
SDL_PauseAudio(false); // Start playback!
soundInitialized = true;
WriteLog("Sound: Successfully initialized.\n");
+
+#ifdef WRITE_OUT_WAVE
+ fp = fopen("./apple2.wav", "wb");
+#endif
}
//
SDL_DestroyMutex(mutex);
SDL_DestroyMutex(mutex2);
WriteLog("Sound: Done.\n");
+
+#ifdef WRITE_OUT_WAVE
+ fclose(fp);
+#endif
}
}
//
// Sound card callback handler
//
-static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length)
+static void SDLSoundCallback(void * userdata, Uint8 * buffer8, int length8)
{
// 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...
SDL_mutexP(mutex2);
- if (soundBufferPos < (uint32)length) // The sound buffer is starved...
+ // Recast this as a 16-bit type...
+ int16 * buffer = (int16 *)buffer8;
+ uint32 length = (uint32)length8 / 2;
+
+ if (soundBufferPos < length) // The sound buffer is starved...
{
-//printf("Sound buffer starved!\n");
-//fflush(stdout);
for(uint32 i=0; i<soundBufferPos; i++)
buffer[i] = soundBuffer[i];
// Fill buffer with last value
- memset(buffer + soundBufferPos, (uint8)(speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]), length - soundBufferPos);
+// memset(buffer + soundBufferPos, (uint8)sample, length - soundBufferPos);
+ for(uint32 i=soundBufferPos; i<length; i++)
+ buffer[i] = (uint16)sample;
soundBufferPos = 0; // Reset soundBufferPos to start of buffer...
- sampleBase = 0; // & sampleBase...
-//Ick. This should never happen!
-//Actually, this probably happens a lot. (?)
-// SDL_CondSignal(conditional); // Wake up any threads waiting for the buffer to drain...
-// return; // & bail!
}
else
{
// Fill sound buffer with frame buffered sound
- memcpy(buffer, soundBuffer, length);
+// memcpy(buffer, soundBuffer, length);
+ for(uint32 i=0; i<length; i++)
+ buffer[i] = soundBuffer[i];
soundBufferPos -= length;
- sampleBase -= length;
// Move current buffer down to start
for(uint32 i=0; i<soundBufferPos; i++)
soundBuffer[i] = soundBuffer[length + i];
}
- // Update our sample position
- samplePosition += length;
// Free the mutex...
SDL_mutexV(mutex2);
// Wake up any threads waiting for the buffer to drain...
the time position back (or copies data down from what it took out)
*/
-void ToggleSpeaker(uint64 elapsedCycles)
+void HandleBuffer(uint64 elapsedCycles)
{
- if (!soundInitialized)
- return;
-
+ // Step 1: Calculate delta time
uint64 deltaCycles = elapsedCycles - lastToggleCycles;
-#if 0
-if (time > 95085)//(time & 0x80000000)
-{
- WriteLog("ToggleSpeaker() given bad time value: %08X (%u)!\n", time, time);
-// fflush(stdout);
-}
-#endif
-
- // 1.024 MHz / 60 = 17066.6... cycles (23.2199 cycles per sample)
- // Need the last frame position in order to calculate correctly...
- // (or do we?)
-
-// SDL_LockAudio();
- SDL_mutexP(mutex2);
-// uint32 currentPos = sampleBase + (uint32)((double)elapsedCycles / CYCLES_PER_SAMPLE);
+ // Step 2: Calculate new buffer position
uint32 currentPos = (uint32)((double)deltaCycles / CYCLES_PER_SAMPLE);
-/*
-The problem:
-
- ______ | ______________ | ______
-____| | | |_______
-
-Speaker is toggled, then not toggled for a while. How to find buffer position in the
-last frame?
-
-IRQ buffer len is 1024.
-
-Could check current CPU clock, take delta. If delta > 1024, then ...
-
-Could add # of cycles in IRQ to lastToggleCycles, then currentPos will be guaranteed
-to fall within acceptable limits.
-*/
+ // Step 3: Make sure there's room for it
+ // We need to lock since we touch both soundBuffer and soundBufferPos
+ SDL_mutexP(mutex2);
+ while ((soundBufferPos + currentPos) > (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...
+ }
+ // Step 4: Backfill and adjust lastToggleCycles
+ // currentPos is position from "zero" or soundBufferPos...
+ currentPos += soundBufferPos;
- if (currentPos > SOUND_BUFFER_SIZE - 1)
- {
-#if 0
-WriteLog("ToggleSpeaker() about to go into spinlock at time: %08X (%u) (sampleBase=%u)!\n", time, time, sampleBase);
+#ifdef WRITE_OUT_WAVE
+ uint32 sbpSave = soundBufferPos;
#endif
-// Still hanging on this spinlock...
-// That could be because the "time" value is too high and so the buffer will NEVER be
-// empty enough...
-// Now that we're using a conditional, it seems to be working OK--though not perfectly...
-/*
-ToggleSpeaker() about to go into spinlock at time: 00004011 (16401) (sampleBase=3504)!
-16401 -> 706 samples, 3504 + 706 = 4210
-
-And it still thrashed the sound even though it didn't run into a spinlock...
+ // Backfill with current toggle state
+ while (soundBufferPos < currentPos)
+ soundBuffer[soundBufferPos++] = (uint16)sample;
-Seems like it's OK now that I've fixed the buffer-less-than-length bug...
-*/
-// SDL_UnlockAudio();
-// SDL_CondWait(conditional, mutex);
-// SDL_LockAudio();
-// Hm.
-SDL_mutexV(mutex2);//Release it so sound thread can get it,
-SDL_CondWait(conditional, mutex);//Sleep/wait for the sound thread
-SDL_mutexP(mutex2);//Re-lock it until we're done with it...
-
- currentPos = sampleBase + (uint32)((double)elapsedCycles / CYCLES_PER_SAMPLE);
-#if 0
-WriteLog("--> after spinlock (sampleBase=%u)...\n", sampleBase);
+#ifdef WRITE_OUT_WAVE
+ fwrite(&soundBuffer[sbpSave], sizeof(int16), currentPos - sbpSave, fp);
#endif
- }
- int8 sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]);
+ SDL_mutexV(mutex2);
+ lastToggleCycles = elapsedCycles;
+}
- while (soundBufferPos < currentPos)
- soundBuffer[soundBufferPos++] = (uint8)sample;
+void ToggleSpeaker(uint64 elapsedCycles)
+{
+ if (!soundInitialized)
+ return;
- // This is done *after* in case the buffer had a long dead spot (I think...)
+ HandleBuffer(elapsedCycles);
speakerState = !speakerState;
- SDL_mutexV(mutex2);
-// SDL_UnlockAudio();
+ sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]);
}
-void AddToSoundTimeBase(uint32 cycles)
+void AdjustLastToggleCycles(uint64 elapsedCycles)
{
if (!soundInitialized)
return;
+/*
+BOOKKEEPING
-// SDL_LockAudio();
- SDL_mutexP(mutex2);
- sampleBase += (uint32)((double)cycles / CYCLES_PER_SAMPLE);
- SDL_mutexV(mutex2);
-// SDL_UnlockAudio();
+We need to know the following:
+
+ o Where in the sound buffer the base or "zero" time is
+ o At what CPU timestamp the speaker was last toggled
+ NOTE: we keep things "right" by advancing this number every frame, even
+ if nothing happened! That way, we can keep track without having
+ to detect whether or not several frames have gone by without any
+ activity.
+
+How to do it:
+
+Every time the speaker is toggled, we move the base or "zero" time to the
+current spot in the buffer. We also backfill the buffer up to that point with
+the old toggle value. The next time the speaker is toggled, we measure the
+difference in time between the last time it was toggled (the "zero") and now,
+and repeat the cycle.
+
+We handle dead spots by backfilling the buffer with the current toggle value
+every frame--this way we don't have to worry about keeping current time and
+crap like that. So, we have to move the "zero" the right amount, just like
+in ToggleSpeaker(), and backfill only without toggling.
+*/
+ HandleBuffer(elapsedCycles);
}
void VolumeUp(void)
{
// Currently set for 8-bit samples
- if (ampPtr < 8)
+ // Now 16
+ if (ampPtr < 16)
ampPtr++;
}
(This may be due to the lock/unlock sound happening in ToggleSpeaker()...)
*/
-
-
-
-
-
-
projects / apple2 / blobdiff