X-Git-Url: http://shamusworld.gotdns.org/cgi-bin/gitweb.cgi?a=blobdiff_plain;ds=sidebyside;f=src%2Fsound.cpp;h=75f9ba2bae9779c3493752f28867905f59fd0d8d;hb=f36d026c7b8b398b88765ec5b67a3c767fe5fbad;hp=ab7802b8adb4ba9d84518ab3cca2d4db28af3881;hpb=ce9f31494277a11aa9cfcbdb5fe5c303627e626e;p=apple2 diff --git a/src/sound.cpp b/src/sound.cpp old mode 100755 new mode 100644 index ab7802b..75f9ba2 --- a/src/sound.cpp +++ b/src/sound.cpp @@ -1,10 +1,10 @@ // // Sound Interface // -// by James L. Hammons +// by James Hammons // (C) 2005 Underground Software // -// JLH = James L. Hammons +// JLH = James Hammons // // WHO WHEN WHAT // --- ---------- ------------------------------------------------------------ @@ -17,73 +17,96 @@ // 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 // For memset, memcpy -#include +#include #include "log.h" -using namespace std; +// 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; +static SDL_AudioSpec desired, obtained; +static SDL_AudioDeviceID device; 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 uint32 soundBufferPos; -static uint32 sampleBase; +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) { -// 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)... + 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 = 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 = 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 + device = SDL_OpenAudioDevice(NULL, 0, &desired, &obtained, 0); + + if (device == 0) { WriteLog("Sound: Failed to initialize SDL sound.\n"); -// exit(1); return; } conditional = SDL_CreateCond(); mutex = SDL_CreateMutex(); - SDL_mutexP(mutex); // Must lock the mutex for the cond to work properly... -// lastValue = (speakerState ? amplitude : 0xFF - amplitude); + mutex2 = SDL_CreateMutex();// Let's try real signalling... soundBufferPos = 0; - sampleBase = 0; + lastToggleCycles = 0; + sample = desired.silence; // ? wilwok ? yes - SDL_PauseAudio(false); // Start playback! + 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 // @@ -91,143 +114,142 @@ void SoundDone(void) { if (soundInitialized) { - SDL_PauseAudio(true); - SDL_CloseAudio(); + 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 * buffer, int length) +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... - - if (soundBufferPos < (uint32)length) // The sound buffer is starved... + // (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) { -//printf("Sound buffer starved!\n"); -//fflush(stdout); - for(uint32 i=0; i 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= (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... + } -So... I guess what we could do is this: + soundBuffer[soundBufferPos++] = sample; +//WriteLog("WriteSampleToBuffer(): SDL_mutexV(mutex2)\n"); + SDL_mutexV(mutex2); +} --- 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) +void ToggleSpeaker(void) { if (!soundInitialized) return; -#if 0 -if (time > 95085)//(time & 0x80000000) -{ - WriteLog("ToggleSpeaker() given bad time value: %08X (%u)!\n", time, time); -// fflush(stdout); + speakerState = !speakerState; + sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]); } -#endif - -// 1.024 MHz / 60 = 17066.6... cycles (23.2199 cycles per sample) -// Need the last frame position in order to calculate correctly... - - SDL_LockAudio(); - uint8 sample = (speakerState ? amplitude : 0xFF - amplitude); -// uint8 sample = (speakerState ? amplitude : amplitude ^ 0xFF); - uint32 currentPos = sampleBase + (uint32)((double)time / 23.2199); - - if (currentPos > 4095) - { -#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); - -// while (currentPos > 4095) // Spin until buffer empties a bit... - currentPos = sampleBase + (uint32)((double)time / 23.2199); - SDL_LockAudio(); -#if 0 -WriteLog("--> after spinlock (sampleBase=%u)...\n", sampleBase); -#endif - } - while (soundBufferPos < currentPos) - soundBuffer[soundBufferPos++] = sample; - speakerState = !speakerState; - SDL_UnlockAudio(); +void VolumeUp(void) +{ + // Currently set for 16-bit samples + if (ampPtr < 16) + ampPtr++; } -void HandleSoundAtFrameEdge(void) -{ - if (!soundInitialized) - return; - SDL_LockAudio(); - sampleBase += 735; - SDL_UnlockAudio(); -/* uint8 sample = (speakerState ? amplitude : 0xFF - amplitude); +void VolumeDown(void) +{ + if (ampPtr > 0) + ampPtr--; +} -//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;//*/ +uint8_t GetVolume(void) +{ + return ampPtr; } +