// 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 AMPLITUDE (32) // -32 - +32 seems to be plenty loud!
+//#define DEBUG
+
+//#define SAMPLE_RATE (44100.0)
+#define SAMPLE_RATE (48000.0)
+#define SAMPLES_PER_FRAME (SAMPLE_RATE / 60.0)
+// ~ 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)
+//nope, too high #define CYCLES_PER_SAMPLE (960000.0 / SAMPLE_RATE)
+//#define SOUND_BUFFER_SIZE (8192)
+#define SOUND_BUFFER_SIZE (16384)
// Global variables
static SDL_AudioSpec desired;
static bool soundInitialized = false;
-static bool speakerState;
-static uint8 soundBuffer[4096];
+static bool speakerState = false;
+static uint8 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 int8 sample;
+static uint8 ampPtr = 5; // Start with -16 - +16
+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.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;
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;
SDL_CloseAudio();
SDL_DestroyCond(conditional);
SDL_DestroyMutex(mutex);
+ SDL_DestroyMutex(mutex2);
WriteLog("Sound: Done.\n");
}
}
// (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...
+ SDL_mutexP(mutex2);
if (soundBufferPos < (uint32)length) // The sound buffer is starved...
{
buffer[i] = soundBuffer[i];
// Fill buffer with last value
- memset(buffer + soundBufferPos, (uint8)(speakerState ? AMPLITUDE : -AMPLITUDE), length - soundBufferPos);
- soundBufferPos = 0; // Reset soundBufferPos to start of buffer...
+// memset(buffer + soundBufferPos, (uint8)(speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]), length - soundBufferPos);
+ memset(buffer + soundBufferPos, (uint8)sample, 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...
+ // Update our sample position
+ samplePosition += length;
+ // 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 ToggleSpeaker(uint64 elapsedCycles)
{
if (!soundInitialized)
return;
+ uint64 deltaCycles = elapsedCycles - lastToggleCycles;
+
#if 0
if (time > 95085)//(time & 0x80000000)
{
// Need the last frame position in order to calculate correctly...
// (or do we?)
- SDL_LockAudio();
- uint32 currentPos = sampleBase + (uint32)((double)time / 23.2199);
+// SDL_LockAudio();
+ SDL_mutexP(mutex2);
+// uint32 currentPos = sampleBase + (uint32)((double)elapsedCycles / CYCLES_PER_SAMPLE);
+ uint32 currentPos = (uint32)((double)deltaCycles / CYCLES_PER_SAMPLE);
+
+/*
+The problem:
+
+ ______ | ______________ | ______
+____| | | |_______
- if (currentPos > 4095)
+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.
+
+This *should* work, but if the IRQ isn't scheduled & etc, could screw timing up.
+Need to have a way to suspend IRQ thread as well as CPU thread when in the GUI,
+for example
+
+Another method would be to add to lastToggleCycles on every timeslice of the CPU,
+just like we used to.
+
+Or, run the CPU for CYCLES_PER_SAMPLE and take a sample, then copy the buffer over
+at the end of the timeslice. That way, we could just fill the buffer and let the
+IRQ handle draining it. No muss, no fuss.
+*/
+
+
+ if ((soundBufferPos + 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);
- SDL_LockAudio();
- currentPos = sampleBase + (uint32)((double)time / 23.2199);
+// SDL_UnlockAudio();
+// SDL_CondWait(conditional, mutex);
+// SDL_LockAudio();
+// Hm.
+// This might not empty the buffer enough, causing hash and trash. !!! FIX !!!
+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)deltaCycles / CYCLES_PER_SAMPLE);
+ currentPos = (uint32)((double)deltaCycles / CYCLES_PER_SAMPLE);
#if 0
WriteLog("--> after spinlock (sampleBase=%u)...\n", sampleBase);
#endif
}
- int8 sample = (speakerState ? AMPLITUDE : -AMPLITUDE);
+ sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]);
+
+ // currentPos is position from "zero" or soundBufferPos...
+ currentPos += soundBufferPos;
while (soundBufferPos < currentPos)
soundBuffer[soundBufferPos++] = (uint8)sample;
+ // This is done *after* in case the buffer had a long dead spot (I think...)
speakerState = !speakerState;
- SDL_UnlockAudio();
+ sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]);
+ lastToggleCycles = elapsedCycles;
+ SDL_mutexV(mutex2);
+// SDL_UnlockAudio();
}
-void HandleSoundAtFrameEdge(void)
+void AddToSoundTimeBase(uint32 cycles)
{
if (!soundInitialized)
return;
- SDL_LockAudio();
- sampleBase += 735;
- SDL_UnlockAudio();
+// SDL_LockAudio();
+ SDL_mutexP(mutex2);
+ sampleBase += (uint32)((double)cycles / CYCLES_PER_SAMPLE);
+ SDL_mutexV(mutex2);
+// SDL_UnlockAudio();
}
+
+void AdjustLastToggleCycles(uint64 elapsedCycles)
+{
+#if 0
+ if (!soundInitialized)
+ return;
+
+ SDL_mutexP(mutex2);
+ lastToggleCycles += elapsedCycles;
+ SDL_mutexV(mutex2);
+
+// We should also fill the buffer here as well, even if the speaker
+// didn't toggle... !!! FIX !!!
+#else
+/*
+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.
+
+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.
+*/
+#warning "This is VERY similar to ToggleSpeaker(); merge into common function. !!! FIX !!!"
+ if (!soundInitialized)
+ return;
+
+#ifdef DEBUG
+printf("SOUND: AdjustLastToggleCycles() start...\n");
+#endif
+ // Step 1: Calculate delta time
+ uint64 deltaCycles = elapsedCycles - lastToggleCycles;
+
+ // Step 2: Calculate new buffer position
+ uint32 currentPos = (uint32)((double)deltaCycles / CYCLES_PER_SAMPLE);
+
+ // 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))
+ {
+ // Hm.
+ // This might not empty the buffer enough, causing hash and trash. !!! FIX !!! [DONE]
+ 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...
+
+//HMM, this doesn't need to lock or recalculate this value
+// currentPos = (uint32)((double)deltaCycles / CYCLES_PER_SAMPLE);
+ }
+
+ // Step 4: Backfill and adjust lastToggleCycles
+ // currentPos is position from "zero" or soundBufferPos...
+ currentPos += soundBufferPos;
+
+ // Backfill with current toggle state
+ while (soundBufferPos < currentPos)
+ soundBuffer[soundBufferPos++] = (uint8)sample;
+
+ SDL_mutexV(mutex2);
+ lastToggleCycles = elapsedCycles;
+#ifdef DEBUG
+printf("SOUND: AdjustLastToggleCycles() end...\n");
+#endif
+#endif
+}
+
+void VolumeUp(void)
+{
+ // Currently set for 8-bit samples
+ if (ampPtr < 8)
+ 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
+
+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()...)
+*/
+
+
+
+
+
+
+