-//\r
-// Sound Interface\r
-//\r
-// by James L. Hammons\r
-// (C) 2005 Underground Software\r
-//\r
-// JLH = James L. Hammons <jlhamm@acm.org>\r
-//\r
-// WHO WHEN WHAT\r
-// --- ---------- ------------------------------------------------------------\r
-// JLH 12/02/2005 Fixed a problem with sound callback thread signaling the\r
-// main thread\r
-// JLH 12/03/2005 Fixed sound callback dropping samples when the sample buffer\r
-// is shorter than the callback sample buffer\r
-//\r
-\r
-// STILL TO DO:\r
-//\r
-// - Figure out why it's losing samples (Bard's Tale) [DONE]\r
-//\r
-\r
-#include "sound.h"\r
-\r
-#include <string.h> // For memset, memcpy\r
-#include <SDL.h>\r
-#include "log.h"\r
-\r
-using namespace std;\r
-\r
-// Global variables\r
-\r
-\r
-// Local variables\r
-\r
-static SDL_AudioSpec desired;\r
-static bool soundInitialized = false;\r
-static uint8 amplitude = 0x88; // $78 - $88 seems to be plenty loud!\r
-//static uint8 lastValue;\r
-\r
-static bool speakerState;\r
-static uint8 soundBuffer[4096];\r
-static uint32 soundBufferPos;\r
-static uint32 sampleBase;\r
-static SDL_cond * conditional = NULL;\r
-static SDL_mutex * mutex = NULL;\r
-\r
-// Private function prototypes\r
-\r
-static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length);\r
-\r
-//\r
-// Initialize the SDL sound system\r
-//\r
-void SoundInit(void)\r
-{\r
-// To weed out problems for now...\r
-#if 0\r
-return;\r
-#endif\r
-\r
- desired.freq = 44100; // SDL will do conversion on the fly, if it can't get the exact rate. Nice!\r
- desired.format = AUDIO_U8; // This uses the native endian (for portability)...\r
- desired.channels = 1;\r
-// desired.samples = 4096; // Let's try a 4K buffer (can always go lower)\r
- desired.samples = 2048; // Let's try a 2K buffer (can always go lower)\r
- desired.callback = SDLSoundCallback;\r
-\r
- if (SDL_OpenAudio(&desired, NULL) < 0) // NULL means SDL guarantees what we want\r
- {\r
- WriteLog("Sound: Failed to initialize SDL sound.\n");\r
-// exit(1);\r
- return;\r
- }\r
-\r
- conditional = SDL_CreateCond();\r
- mutex = SDL_CreateMutex();\r
- SDL_mutexP(mutex); // Must lock the mutex for the cond to work properly...\r
-// lastValue = (speakerState ? amplitude : 0xFF - amplitude);\r
- soundBufferPos = 0;\r
- sampleBase = 0;\r
-\r
- SDL_PauseAudio(false); // Start playback!\r
- soundInitialized = true;\r
- WriteLog("Sound: Successfully initialized.\n");\r
-}\r
-\r
-//\r
-// Close down the SDL sound subsystem\r
-//\r
-void SoundDone(void)\r
-{\r
- if (soundInitialized)\r
- {\r
- SDL_PauseAudio(true);\r
- SDL_CloseAudio();\r
- SDL_DestroyCond(conditional);\r
- SDL_DestroyMutex(mutex);\r
- WriteLog("Sound: Done.\n");\r
- }\r
-}\r
-\r
-//\r
-// Sound card callback handler\r
-//\r
-static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length)\r
-{\r
- // The sound buffer should only starve when starting which will cause it to\r
- // lag behind the emulation at most by around 1 frame...\r
-\r
- if (soundBufferPos < (uint32)length) // The sound buffer is starved...\r
- {\r
-//printf("Sound buffer starved!\n");\r
-//fflush(stdout);\r
- for(uint32 i=0; i<soundBufferPos; i++)\r
- buffer[i] = soundBuffer[i];\r
- // Fill buffer with last value\r
- uint8 lastValue = (speakerState ? amplitude : 0xFF - amplitude);\r
-// uint8 lastValue = (speakerState ? amplitude : amplitude ^ 0xFF);\r
-// memset(buffer, lastValue, length); // Fill buffer with last value\r
- memset(buffer + soundBufferPos, lastValue, length - soundBufferPos);\r
- soundBufferPos = 0; // Reset soundBufferPos to start of buffer...\r
- sampleBase = 0; // & sampleBase...\r
-//Ick. This should never happen!\r
-SDL_CondSignal(conditional); // Wake up any threads waiting for the buffer to drain...\r
- return; // & bail!\r
- }\r
-\r
- memcpy(buffer, soundBuffer, length); // Fill sound buffer with frame buffered sound\r
- soundBufferPos -= length;\r
- sampleBase -= length;\r
-\r
-// if (soundBufferPos > 0)\r
-// memcpy(soundBuffer, soundBuffer + length, soundBufferPos); // Move current buffer down to start\r
-// memcpy(soundBuffer, soundBuffer + length, length);\r
- // Move current buffer down to start\r
- for(uint32 i=0; i<soundBufferPos; i++)\r
- soundBuffer[i] = soundBuffer[length + i];\r
-\r
-// lastValue = buffer[length - 1];\r
- SDL_CondSignal(conditional); // Wake up any threads waiting for the buffer to drain...\r
-}\r
-\r
-// Need some interface functions here to take care of flipping the\r
-// waveform at the correct time in the sound stream...\r
-\r
-/*\r
-Maybe set up a buffer 1 frame long (44100 / 60 = 735 bytes per frame)\r
-\r
-Hmm. That's smaller than the sound buffer 2048 bytes... (About 2.75 frames needed to fill)\r
-\r
-So... I guess what we could do is this:\r
-\r
--- Execute V65C02 for one frame. The read/writes at I/O address $C030 fill up the buffer\r
- to the current time position.\r
--- The sound callback function copies the pertinent area out of the buffer, resets\r
- the time position back (or copies data down from what it took out)\r
-*/\r
-\r
-void ToggleSpeaker(uint32 time)\r
-{\r
- if (!soundInitialized)\r
- return;\r
-\r
-#if 0\r
-if (time > 95085)//(time & 0x80000000)\r
-{\r
- WriteLog("ToggleSpeaker() given bad time value: %08X (%u)!\n", time, time);\r
-// fflush(stdout);\r
-}\r
-#endif\r
-\r
-// 1.024 MHz / 60 = 17066.6... cycles (23.2199 cycles per sample)\r
-// Need the last frame position in order to calculate correctly...\r
-\r
- SDL_LockAudio();\r
- uint8 sample = (speakerState ? amplitude : 0xFF - amplitude);\r
-// uint8 sample = (speakerState ? amplitude : amplitude ^ 0xFF);\r
- uint32 currentPos = sampleBase + (uint32)((double)time / 23.2199);\r
-\r
- if (currentPos > 4095)\r
- {\r
-#if 0\r
-WriteLog("ToggleSpeaker() about to go into spinlock at time: %08X (%u) (sampleBase=%u)!\n", time, time, sampleBase);\r
-#endif\r
-// Still hanging on this spinlock...\r
-// That could be because the "time" value is too high and so the buffer will NEVER be\r
-// empty enough...\r
-// Now that we're using a conditional, it seems to be working OK--though not perfectly...\r
-/*\r
-ToggleSpeaker() about to go into spinlock at time: 00004011 (16401) (sampleBase=3504)!\r
-16401 -> 706 samples, 3504 + 706 = 4210\r
-\r
-And it still thrashed the sound even though it didn't run into a spinlock...\r
-\r
-Seems like it's OK now that I've fixed the buffer-less-than-length bug...\r
-*/\r
- SDL_UnlockAudio();\r
- SDL_CondWait(conditional, mutex);\r
-\r
-// while (currentPos > 4095) // Spin until buffer empties a bit...\r
- currentPos = sampleBase + (uint32)((double)time / 23.2199);\r
- SDL_LockAudio();\r
-#if 0\r
-WriteLog("--> after spinlock (sampleBase=%u)...\n", sampleBase);\r
-#endif\r
- }\r
-\r
- while (soundBufferPos < currentPos)\r
- soundBuffer[soundBufferPos++] = sample;\r
-\r
- speakerState = !speakerState;\r
- SDL_UnlockAudio();\r
-}\r
-\r
-void HandleSoundAtFrameEdge(void)\r
-{\r
- if (!soundInitialized)\r
- return;\r
-\r
- SDL_LockAudio();\r
- sampleBase += 735;\r
- SDL_UnlockAudio();\r
-/* uint8 sample = (speakerState ? amplitude : 0xFF - amplitude);\r
-\r
-//This shouldn't happen (buffer overflow), but it seems like it *is* happening...\r
- if (sampleBase >= 4096)\r
-// sampleBase = 4095;\r
-//Kludge, for now... Until I can figure out why it's still stomping on the buffer...\r
- sampleBase = 0;\r
-\r
- while (soundBufferPos < sampleBase)\r
- soundBuffer[soundBufferPos++] = sample;//*/\r
-}\r
+//
+// Sound Interface
+//
+// by James L. Hammons
+// (C) 2005 Underground Software
+//
+// JLH = James L. Hammons <jlhamm@acm.org>
+//
+// WHO WHEN WHAT
+// --- ---------- ------------------------------------------------------------
+// JLH 12/02/2005 Fixed a problem with sound callback thread signaling the
+// main thread
+// JLH 12/03/2005 Fixed sound callback dropping samples when the sample buffer
+// is shorter than the callback sample buffer
+//
+
+// STILL TO DO:
+//
+// - Figure out why it's losing samples (Bard's Tale) [DONE]
+//
+
+#include "sound.h"
+
+#include <string.h> // For memset, memcpy
+#include <SDL.h>
+#include "log.h"
+
+
+#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
+
+
+// Local variables
+
+static SDL_AudioSpec desired;
+static bool soundInitialized = false;
+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
+
+static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length);
+
+//
+// Initialize the SDL sound system
+//
+void SoundInit(void)
+{
+#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.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
+ {
+ 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;
+
+ SDL_PauseAudio(false); // Start playback!
+ soundInitialized = true;
+ WriteLog("Sound: Successfully initialized.\n");
+}
+
+//
+// Close down the SDL sound subsystem
+//
+void SoundDone(void)
+{
+ if (soundInitialized)
+ {
+ SDL_PauseAudio(true);
+ SDL_CloseAudio();
+ SDL_DestroyCond(conditional);
+ SDL_DestroyMutex(mutex);
+ SDL_DestroyMutex(mutex2);
+ WriteLog("Sound: Done.\n");
+ }
+}
+
+//
+// Sound card callback handler
+//
+static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length)
+{
+ // 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...
+ {
+//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);
+ 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);
+ soundBufferPos -= length;
+ sampleBase -= length;
+
+ // Move current buffer down to start
+ for(uint32 i=0; i<soundBufferPos; i++)
+ soundBuffer[i] = soundBuffer[length + i];
+ }
+
+ // 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
+// waveform at the correct time in the sound stream...
+
+/*
+Maybe set up a buffer 1 frame long (44100 / 60 = 735 bytes per frame)
+
+Hmm. That's smaller than the sound buffer 2048 bytes... (About 2.75 frames needed to fill)
+
+So... I guess what we could do is this:
+
+-- Execute V65C02 for one frame. The read/writes at I/O address $C030 fill up the buffer
+ to the current time position.
+-- The sound callback function copies the pertinent area out of the buffer, resets
+ the time position back (or copies data down from what it took out)
+*/
+
+void ToggleSpeaker(uint32 time)
+{
+ if (!soundInitialized)
+ return;
+
+#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)time / CYCLES_PER_SAMPLE);
+
+ 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);
+#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...
+
+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 / 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++] = (uint8)sample;
+
+ // This is done *after* in case the buffer had a long dead spot (I think...)
+ speakerState = !speakerState;
+ SDL_mutexV(mutex2);
+// SDL_UnlockAudio();
+}
+
+void AddToSoundTimeBase(uint32 cycles)
+{
+ if (!soundInitialized)
+ return;
+
+// 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()...)
+*/
+
+
+
+
+
+
+
projects / apple2 / blobdiff