-//\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 Hammons
+// (C) 2005 Underground Software
+//
+// JLH = James 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]
+// - Figure out why it's playing too fast [DONE]
+//
+
+#include "sound.h"
+
+#include <string.h> // For memset, memcpy
+#include <SDL2/SDL.h>
+#include "log.h"
+
+// Useful defines
+
+//#define DEBUG
+//#define WRITE_OUT_WAVE
+
+//#define SAMPLE_RATE (44100.0)
+#define SAMPLE_RATE (48000.0)
+#define SAMPLES_PER_FRAME (SAMPLE_RATE / 60.0)
+#define CYCLES_PER_SAMPLE (1024000.0 / SAMPLE_RATE)
+//#define SOUND_BUFFER_SIZE (8192)
+#define SOUND_BUFFER_SIZE (32768)
+
+// Global variables
+
+
+// Local variables
+
+static SDL_AudioSpec desired, obtained;
+static SDL_AudioDeviceID device;
+static bool soundInitialized = false;
+static bool speakerState = false;
+static int16_t soundBuffer[SOUND_BUFFER_SIZE];
+static uint32_t soundBufferPos;
+static uint64_t lastToggleCycles;
+static SDL_cond * conditional = NULL;
+static SDL_mutex * mutex = NULL;
+static SDL_mutex * mutex2 = NULL;
+static int16_t sample;
+static uint8_t ampPtr = 12; // Start with -2047 - +2047
+static int16_t 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
+
+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
+ SDL_zero(desired);
+ desired.freq = SAMPLE_RATE; // SDL will do conversion on the fly, if it can't get the exact rate. Nice!
+ desired.format = AUDIO_S16SYS; // This uses the native endian (for portability)...
+ desired.channels = 1;
+ desired.samples = 512; // Let's try a 1/2K buffer (can always go lower)
+ desired.callback = SDLSoundCallback;
+
+ device = SDL_OpenAudioDevice(NULL, 0, &desired, &obtained, 0);
+
+ if (device == 0)
+ {
+ WriteLog("Sound: Failed to initialize SDL sound.\n");
+ return;
+ }
+
+ conditional = SDL_CreateCond();
+ mutex = SDL_CreateMutex();
+ mutex2 = SDL_CreateMutex();// Let's try real signalling...
+ soundBufferPos = 0;
+ lastToggleCycles = 0;
+ sample = desired.silence; // ? wilwok ? yes
+
+ SDL_PauseAudioDevice(device, 0); // Start playback!
+ soundInitialized = true;
+ WriteLog("Sound: Successfully initialized.\n");
+
+#ifdef WRITE_OUT_WAVE
+ fp = fopen("./apple2.wav", "wb");
+#endif
+}
+
+
+//
+// Close down the SDL sound subsystem
+//
+void SoundDone(void)
+{
+ if (soundInitialized)
+ {
+ SDL_PauseAudioDevice(device, 1);
+ SDL_CloseAudioDevice(device);
+ SDL_DestroyCond(conditional);
+ SDL_DestroyMutex(mutex);
+ SDL_DestroyMutex(mutex2);
+ WriteLog("Sound: Done.\n");
+
+#ifdef WRITE_OUT_WAVE
+ fclose(fp);
+#endif
+ }
+}
+
+
+void SoundPause(void)
+{
+ if (soundInitialized)
+ SDL_PauseAudioDevice(device, 1);
+}
+
+
+void SoundResume(void)
+{
+ if (soundInitialized)
+ SDL_PauseAudioDevice(device, 0);
+}
+
+
+//
+// Sound card callback handler
+//
+static void SDLSoundCallback(void * /*userdata*/, Uint8 * buffer8, int length8)
+{
+//WriteLog("SDLSoundCallback(): begin (soundBufferPos=%i)\n", soundBufferPos);
+ // 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...
+//WriteLog("SDLSoundCallback: soundBufferPos = %i\n", soundBufferPos);
+ SDL_mutexP(mutex2);
+
+ // Recast this as a 16-bit type...
+ int16_t * buffer = (int16_t *)buffer8;
+ uint32_t length = (uint32_t)length8 / 2;
+
+//WriteLog("SDLSoundCallback(): filling buffer...\n");
+ if (soundBufferPos < length)
+ {
+ // The sound buffer is starved...
+ for(uint32_t i=0; i<soundBufferPos; i++)
+ buffer[i] = soundBuffer[i];
+
+ // Fill buffer with last value
+ for(uint32_t i=soundBufferPos; i<length; i++)
+ buffer[i] = sample;
+
+ // Reset soundBufferPos to start of buffer...
+ soundBufferPos = 0;
+ }
+ else
+ {
+ // Fill sound buffer with frame buffered sound
+ for(uint32_t i=0; i<length; i++)
+ buffer[i] = soundBuffer[i];
+
+ soundBufferPos -= length;
+
+ // Move current buffer down to start
+ for(uint32_t i=0; i<soundBufferPos; i++)
+ soundBuffer[i] = soundBuffer[length + i];
+ }
+
+ // Free the mutex...
+//WriteLog("SDLSoundCallback(): SDL_mutexV(mutex2)\n");
+ SDL_mutexV(mutex2);
+ // Wake up any threads waiting for the buffer to drain...
+ SDL_CondSignal(conditional);
+//WriteLog("SDLSoundCallback(): end\n");
+}
+
+
+// This is called by the main CPU thread every ~21.333 cycles.
+void WriteSampleToBuffer(void)
+{
+//WriteLog("WriteSampleToBuffer(): SDL_mutexP(mutex2)\n");
+ SDL_mutexP(mutex2);
+
+ // This should almost never happen, but, if it does...
+ while (soundBufferPos >= (SOUND_BUFFER_SIZE - 1))
+ {
+//WriteLog("WriteSampleToBuffer(): Waiting for sound thread. soundBufferPos=%i, SOUNDBUFFERSIZE-1=%i\n", soundBufferPos, 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...
+ }
+
+ soundBuffer[soundBufferPos++] = sample;
+//WriteLog("WriteSampleToBuffer(): SDL_mutexV(mutex2)\n");
+ SDL_mutexV(mutex2);
+}
+
+
+void ToggleSpeaker(void)
+{
+ if (!soundInitialized)
+ return;
+
+ speakerState = !speakerState;
+ sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]);
+}
+
+
+void VolumeUp(void)
+{
+ // Currently set for 16-bit samples
+ if (ampPtr < 16)
+ ampPtr++;
+}
+
+
+void VolumeDown(void)
+{
+ if (ampPtr > 0)
+ ampPtr--;
+}
+
+
+uint8_t GetVolume(void)
+{
+ return ampPtr;
+}
+
projects / apple2 / blobdiff