#include <SDL.h>
#include "log.h"
-using namespace std;
+
+#define SAMPLE_RATE (44100.0)
+#define SAMPLES_PER_FRAME (SAMPLE_RATE / 60.0)
+#define CYCLES_PER_SAMPLE (1024000.0 / SAMPLE_RATE)
+#define SOUND_BUFFER_SIZE (8192)
+//#define AMPLITUDE (16) // -32 - +32 seems to be plenty loud!
// Global variables
static SDL_AudioSpec desired;
static bool soundInitialized = false;
-static uint8 amplitude = 0x88; // $78 - $88 seems to be plenty loud!
-//static uint8 lastValue;
-
-static bool speakerState;
-static uint8 soundBuffer[4096];
+static bool speakerState = false;
+static uint8 soundBuffer[SOUND_BUFFER_SIZE];
static uint32 soundBufferPos;
static uint32 sampleBase;
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 };
// Private function prototypes
//
void SoundInit(void)
{
-// To weed out problems for now...
#if 0
+// To weed out problems for now...
return;
#endif
- desired.freq = 44100; // SDL will do conversion on the fly, if it can't get the exact rate. Nice!
- desired.format = AUDIO_U8; // 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_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 = 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
{
WriteLog("Sound: Failed to initialize SDL sound.\n");
-// exit(1);
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...
-// lastValue = (speakerState ? amplitude : 0xFF - amplitude);
soundBufferPos = 0;
sampleBase = 0;
SDL_CloseAudio();
SDL_DestroyCond(conditional);
SDL_DestroyMutex(mutex);
+ SDL_DestroyMutex(mutex2);
WriteLog("Sound: Done.\n");
}
}
{
// 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!)
+
+ // Let's try using a mutex for shared resource consumption...
+ SDL_mutexP(mutex2);
if (soundBufferPos < (uint32)length) // The sound buffer is starved...
{
//fflush(stdout);
for(uint32 i=0; i<soundBufferPos; i++)
buffer[i] = soundBuffer[i];
+
// Fill buffer with last value
- uint8 lastValue = (speakerState ? amplitude : 0xFF - amplitude);
-// uint8 lastValue = (speakerState ? amplitude : amplitude ^ 0xFF);
-// memset(buffer, lastValue, length); // Fill buffer with last value
- memset(buffer + soundBufferPos, lastValue, length - soundBufferPos);
+ memset(buffer + soundBufferPos, (uint8)(speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]), length - soundBufferPos);
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!
+//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);
+ soundBufferPos -= length;
+ sampleBase -= 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
}
#endif
-// 1.024 MHz / 60 = 17066.6... cycles (23.2199 cycles per sample)
-// Need the last frame position in order to calculate correctly...
+ // 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();
- uint8 sample = (speakerState ? amplitude : 0xFF - amplitude);
-// uint8 sample = (speakerState ? amplitude : amplitude ^ 0xFF);
- uint32 currentPos = sampleBase + (uint32)((double)time / 23.2199);
+// SDL_LockAudio();
+ SDL_mutexP(mutex2);
+ uint32 currentPos = sampleBase + (uint32)((double)time / CYCLES_PER_SAMPLE);
- if (currentPos > 4095)
+ 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);
Seems like it's OK now that I've fixed the buffer-less-than-length bug...
*/
- SDL_UnlockAudio();
- SDL_CondWait(conditional, mutex);
-
-// while (currentPos > 4095) // Spin until buffer empties a bit...
- currentPos = sampleBase + (uint32)((double)time / 23.2199);
- SDL_LockAudio();
+// SDL_UnlockAudio();
+// SDL_CondWait(conditional, mutex);
+// SDL_LockAudio();
+ currentPos = sampleBase + (uint32)((double)time / CYCLES_PER_SAMPLE);
#if 0
WriteLog("--> after spinlock (sampleBase=%u)...\n", sampleBase);
#endif
}
+ int8 sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]);
+
while (soundBufferPos < currentPos)
- soundBuffer[soundBufferPos++] = sample;
+ soundBuffer[soundBufferPos++] = (uint8)sample;
+ // This is done *after* in case the buffer had a long dead spot (I think...)
speakerState = !speakerState;
- SDL_UnlockAudio();
+ SDL_mutexV(mutex2);
+// SDL_UnlockAudio();
}
-void HandleSoundAtFrameEdge(void)
+void AddToSoundTimeBase(uint32 cycles)
{
if (!soundInitialized)
return;
- SDL_LockAudio();
- sampleBase += 735;
- SDL_UnlockAudio();
-/* uint8 sample = (speakerState ? amplitude : 0xFF - amplitude);
+// SDL_LockAudio();
+ SDL_mutexP(mutex2);
+ sampleBase += (uint32)((double)cycles / CYCLES_PER_SAMPLE);
+ SDL_mutexV(mutex2);
+// SDL_UnlockAudio();
+}
+
+/*
+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
+
+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()...)
+*/
+
+
+
+
+
-//This shouldn't happen (buffer overflow), but it seems like it *is* happening...
- if (sampleBase >= 4096)
-// sampleBase = 4095;
-//Kludge, for now... Until I can figure out why it's still stomping on the buffer...
- sampleBase = 0;
- while (soundBufferPos < sampleBase)
- soundBuffer[soundBufferPos++] = sample;//*/
-}
projects / apple2 / blobdiff