// 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 SAMPLE_RATE (44100.0)
+//#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)
-#define CYCLES_PER_SAMPLE ((1024000.0 / 60.0) / (SAMPLES_PER_FRAME))
-#define SOUND_BUFFER_SIZE 8192
-#define AMPLITUDE (32) // -32 - +32 seems to be plenty loud!
+// This works for AppleWin but not here... ??? WHY ???
+// ~ 21
+#define CYCLES_PER_SAMPLE (1024000.0 / SAMPLE_RATE)
+// ~ 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;
-static uint8 soundBuffer[SOUND_BUFFER_SIZE];
+static bool speakerState = false;
+static int16 soundBuffer[SOUND_BUFFER_SIZE];
static uint32 soundBufferPos;
-static uint32 sampleBase;
+static uint64 lastToggleCycles;
static SDL_cond * conditional = NULL;
static SDL_mutex * mutex = NULL;
+static SDL_mutex * mutex2 = NULL;
+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
//
void SoundInit(void)
{
-// To weed out problems for now...
#if 0
+// To weed out problems for now...
return;
#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.samples = 1024; // Let's try a 1K buffer (can always go lower)
+ desired.samples = 512; // Let's try a 1/2K 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();
- SDL_mutexP(mutex); // Must lock the mutex for the cond to work properly...
+ mutex2 = SDL_CreateMutex();// Let's try real signalling...
soundBufferPos = 0;
- sampleBase = 0;
+ lastToggleCycles = 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_CloseAudio();
SDL_DestroyCond(conditional);
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...)
- if (soundBufferPos < (uint32)length) // The sound buffer is starved...
+ // Let's try using a mutex for shared resource consumption...
+//Actually, I think Lock/UnlockAudio() does this already...
+ SDL_mutexP(mutex2);
+
+ // 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 : -AMPLITUDE), 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!
-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);
+ for(uint32 i=0; i<length; i++)
+ buffer[i] = soundBuffer[i];
+ soundBufferPos -= length;
- memcpy(buffer, soundBuffer, length); // Fill sound buffer with frame buffered sound
- soundBufferPos -= length;
- sampleBase -= length;
-
-// if (soundBufferPos > 0)
-// memcpy(soundBuffer, soundBuffer + length, soundBufferPos); // Move current buffer down to start
-// memcpy(soundBuffer, soundBuffer + length, length);
- // Move current buffer down to start
- for(uint32 i=0; i<soundBufferPos; i++)
- soundBuffer[i] = soundBuffer[length + i];
+ // Move current buffer down to start
+ for(uint32 i=0; i<soundBufferPos; i++)
+ soundBuffer[i] = soundBuffer[length + i];
+ }
-// lastValue = buffer[length - 1];
- SDL_CondSignal(conditional); // Wake up any threads waiting for the buffer to drain...
+ // Free the mutex...
+ SDL_mutexV(mutex2);
+ // Wake up any threads waiting for the buffer to drain...
+ SDL_CondSignal(conditional);
}
// Need some interface functions here to take care of flipping the
the time position back (or copies data down from what it took out)
*/
-void ToggleSpeaker(uint32 time)
+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
+ // Step 2: Calculate new buffer position
+ uint32 currentPos = (uint32)((double)deltaCycles / CYCLES_PER_SAMPLE);
- // 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?)
+ // 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...
+ }
- SDL_LockAudio();
- uint32 currentPos = sampleBase + (uint32)((double)time / CYCLES_PER_SAMPLE);
+ // 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();
- currentPos = sampleBase + (uint32)((double)time / 23.2199);
-#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 : -AMPLITUDE);
+ SDL_mutexV(mutex2);
+ lastToggleCycles = elapsedCycles;
+}
- while (soundBufferPos < currentPos)
- soundBuffer[soundBufferPos++] = (uint8)sample;
+void ToggleSpeaker(uint64 elapsedCycles)
+{
+ if (!soundInitialized)
+ return;
+ HandleBuffer(elapsedCycles);
speakerState = !speakerState;
- SDL_UnlockAudio();
+ sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]);
}
-void HandleSoundAtFrameEdge(void)
+void AdjustLastToggleCycles(uint64 elapsedCycles)
{
if (!soundInitialized)
return;
+/*
+BOOKKEEPING
+
+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.
- SDL_LockAudio();
- sampleBase += SAMPLES_PER_FRAME;
- SDL_UnlockAudio();
+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
+ // Now 16
+ if (ampPtr < 16)
+ ampPtr++;
+}
+
+void VolumeDown(void)
+{
+ if (ampPtr > 0)
+ ampPtr--;
+}
+
+uint8 GetVolume(void)
+{
+ return ampPtr;
}
/*
+HOW IT WORKS
+
+the main thread adds the amount of cpu time elapsed to samplebase. togglespeaker uses
+samplebase + current cpu time to find appropriate spot in buffer. it then fills the
+buffer up to the current time with the old toggle value before flipping it. the sound
+irq takes what it needs from the sound buffer and then adjusts both the buffer and
+samplebase back the appropriate amount.
+
+
A better way might be as follows:
-Keep timestamp array of speaker toggle times. In the sound routine, unpack as many as will fit
-into the given buffer and keep going. Have the toggle function check to see if the buffer is full,
-and if it is, way for a signal from the interrupt that there's room for more. Can keep a circular
-buffer. Also, would need a timestamp buffer on the order of 2096 samples *in theory* could toggle
-each sample
+Keep timestamp array of speaker toggle times. In the sound routine, unpack as many as will
+fit into the given buffer and keep going. Have the toggle function check to see if the
+buffer is full, and if it is, way for a signal from the interrupt that there's room for
+more. Can keep a circular buffer. Also, would need a timestamp buffer on the order of 2096
+samples *in theory* could toggle each sample
-Instead of a timestamp, just keep a delta. That way, don't need to deal with wrapping and all that
-(though the timestamp could wrap--need to check into that)
+Instead of a timestamp, just keep a delta. That way, don't need to deal with wrapping and
+all that (though the timestamp could wrap--need to check into that)
Need to consider corner cases where a sound IRQ happens but no speaker toggle happened.
-*/
+If (delta > SAMPLES_PER_FRAME) then
+
+Here's the relevant cases:
+
+delta < SAMPLES_PER_FRAME -> Change happened within this time frame, so change buffer
+frame came and went, no change -> fill buffer with last value
+How to detect: Have bool bufferWasTouched = true when ToggleSpeaker() is called.
+Clear bufferWasTouched each frame.
+Two major cases here:
+ o Buffer is touched on current frame
+ o Buffer is untouched on current frame
+
+In the first case, it doesn't matter too much if the previous frame was touched or not,
+we don't really care except in finding the correct spot in the buffer to put our change
+in. In the second case, we need to tell the IRQ that nothing happened and to continue
+to output the same value.
+
+SO: How to synchronize the regular frame buffer with the IRQ buffer?
+
+What happens:
+ Sound IRQ --> Every 1024 sample period (@ 44.1 KHz = 0.0232s)
+ Emulation --> Render a frame --> 1/60 sec --> 735 samples
+ --> sound buffer is filled
+
+Since the emulation is faster than the SIRQ the sound buffer should fill up
+prior to dumping it to the sound card.
+
+Problem is this: If silence happens for a long time then ToggleSpeaker is never
+called and the sound buffer has stale data; at least until soundBufferPos goes to
+zero and stays there...
+
+BUT this should be handled correctly by toggling the speaker value *after* filling
+the sound buffer...
+
+Still getting random clicks when running...
+(This may be due to the lock/unlock sound happening in ToggleSpeaker()...)
+*/
projects / apple2 / blobdiff