2007-05-29 03:06:33 +00:00
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//
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// Sound Interface
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//
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// by James L. Hammons
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// (C) 2005 Underground Software
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//
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// JLH = James L. Hammons <jlhamm@acm.org>
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//
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// WHO WHEN WHAT
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// --- ---------- ------------------------------------------------------------
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// JLH 12/02/2005 Fixed a problem with sound callback thread signaling the
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// main thread
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// JLH 12/03/2005 Fixed sound callback dropping samples when the sample buffer
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// is shorter than the callback sample buffer
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//
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// STILL TO DO:
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//
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// - Figure out why it's losing samples (Bard's Tale) [DONE]
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//
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#include "sound.h"
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#include <string.h> // For memset, memcpy
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#include <SDL.h>
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#include "log.h"
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2009-01-19 16:09:37 +00:00
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2009-01-19 20:00:34 +00:00
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#define SAMPLE_RATE (44100.0)
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#define SAMPLES_PER_FRAME (SAMPLE_RATE / 60.0)
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2009-01-22 22:12:49 +00:00
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#define CYCLES_PER_SAMPLE (1024000.0 / SAMPLE_RATE)
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#define SOUND_BUFFER_SIZE (8192)
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#define AMPLITUDE (32) // -32 - +32 seems to be plenty loud!
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2007-05-29 03:06:33 +00:00
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// Global variables
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// Local variables
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static SDL_AudioSpec desired;
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static bool soundInitialized = false;
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2009-01-22 22:12:49 +00:00
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static bool speakerState = false;
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2009-01-19 20:00:34 +00:00
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static uint8 soundBuffer[SOUND_BUFFER_SIZE];
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2007-05-29 03:06:33 +00:00
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static uint32 soundBufferPos;
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static uint32 sampleBase;
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static SDL_cond * conditional = NULL;
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static SDL_mutex * mutex = NULL;
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2009-01-22 22:12:49 +00:00
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static SDL_mutex * mutex2 = NULL;
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2007-05-29 03:06:33 +00:00
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// Private function prototypes
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static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length);
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//
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// Initialize the SDL sound system
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//
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void SoundInit(void)
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{
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// To weed out problems for now...
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#if 0
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return;
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#endif
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2009-01-19 20:00:34 +00:00
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desired.freq = SAMPLE_RATE; // SDL will do conversion on the fly, if it can't get the exact rate. Nice!
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2009-01-19 16:09:37 +00:00
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desired.format = AUDIO_S8; // This uses the native endian (for portability)...
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// desired.format = AUDIO_S16SYS; // This uses the native endian (for portability)...
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2007-05-29 03:06:33 +00:00
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desired.channels = 1;
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// desired.samples = 4096; // Let's try a 4K buffer (can always go lower)
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2009-01-19 16:09:37 +00:00
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// desired.samples = 2048; // Let's try a 2K buffer (can always go lower)
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desired.samples = 1024; // Let's try a 1K buffer (can always go lower)
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2007-05-29 03:06:33 +00:00
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desired.callback = SDLSoundCallback;
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if (SDL_OpenAudio(&desired, NULL) < 0) // NULL means SDL guarantees what we want
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{
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WriteLog("Sound: Failed to initialize SDL sound.\n");
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return;
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}
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conditional = SDL_CreateCond();
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mutex = SDL_CreateMutex();
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2009-01-22 22:12:49 +00:00
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mutex2 = SDL_CreateMutex();// Let's try real signalling...
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2007-05-29 03:06:33 +00:00
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SDL_mutexP(mutex); // Must lock the mutex for the cond to work properly...
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soundBufferPos = 0;
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sampleBase = 0;
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SDL_PauseAudio(false); // Start playback!
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soundInitialized = true;
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WriteLog("Sound: Successfully initialized.\n");
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}
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//
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// Close down the SDL sound subsystem
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//
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void SoundDone(void)
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{
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if (soundInitialized)
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{
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SDL_PauseAudio(true);
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SDL_CloseAudio();
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SDL_DestroyCond(conditional);
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SDL_DestroyMutex(mutex);
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2009-01-22 22:12:49 +00:00
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SDL_DestroyMutex(mutex2);
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2007-05-29 03:06:33 +00:00
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WriteLog("Sound: Done.\n");
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}
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}
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//
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// Sound card callback handler
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//
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static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length)
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{
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// The sound buffer should only starve when starting which will cause it to
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// lag behind the emulation at most by around 1 frame...
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2009-01-19 16:09:37 +00:00
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// (Actually, this should never happen since we fill the buffer beforehand.)
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// (But, then again, if the sound hasn't been toggled for a while, then this
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// makes perfect sense as the buffer won't have been filled at all!)
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2007-05-29 03:06:33 +00:00
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2009-01-22 22:12:49 +00:00
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// Let's try using a mutex for shared resource consumption...
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SDL_mutexP(mutex2);
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2007-05-29 03:06:33 +00:00
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if (soundBufferPos < (uint32)length) // The sound buffer is starved...
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{
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//printf("Sound buffer starved!\n");
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//fflush(stdout);
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for(uint32 i=0; i<soundBufferPos; i++)
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buffer[i] = soundBuffer[i];
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2009-01-19 16:09:37 +00:00
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2007-05-29 03:06:33 +00:00
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// Fill buffer with last value
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2009-01-19 16:09:37 +00:00
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memset(buffer + soundBufferPos, (uint8)(speakerState ? AMPLITUDE : -AMPLITUDE), length - soundBufferPos);
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2007-05-29 03:06:33 +00:00
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soundBufferPos = 0; // Reset soundBufferPos to start of buffer...
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sampleBase = 0; // & sampleBase...
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//Ick. This should never happen!
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2009-01-22 22:12:49 +00:00
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//Actually, this probably happens a lot. (?)
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// SDL_CondSignal(conditional); // Wake up any threads waiting for the buffer to drain...
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// return; // & bail!
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2007-05-29 03:06:33 +00:00
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}
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2009-01-22 22:12:49 +00:00
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else
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{
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// Fill sound buffer with frame buffered sound
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memcpy(buffer, soundBuffer, length);
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soundBufferPos -= length;
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sampleBase -= length;
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2007-05-29 03:06:33 +00:00
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2009-01-22 22:12:49 +00:00
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// Move current buffer down to start
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for(uint32 i=0; i<soundBufferPos; i++)
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soundBuffer[i] = soundBuffer[length + i];
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}
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2007-05-29 03:06:33 +00:00
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2009-01-22 22:12:49 +00:00
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// Free the mutex...
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SDL_mutexV(mutex2);
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// Wake up any threads waiting for the buffer to drain...
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// SDL_CondSignal(conditional);
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2007-05-29 03:06:33 +00:00
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}
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// Need some interface functions here to take care of flipping the
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// waveform at the correct time in the sound stream...
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/*
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Maybe set up a buffer 1 frame long (44100 / 60 = 735 bytes per frame)
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Hmm. That's smaller than the sound buffer 2048 bytes... (About 2.75 frames needed to fill)
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So... I guess what we could do is this:
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-- Execute V65C02 for one frame. The read/writes at I/O address $C030 fill up the buffer
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to the current time position.
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-- The sound callback function copies the pertinent area out of the buffer, resets
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the time position back (or copies data down from what it took out)
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*/
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void ToggleSpeaker(uint32 time)
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{
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if (!soundInitialized)
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return;
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#if 0
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if (time > 95085)//(time & 0x80000000)
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{
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WriteLog("ToggleSpeaker() given bad time value: %08X (%u)!\n", time, time);
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// fflush(stdout);
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}
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#endif
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2009-01-19 16:09:37 +00:00
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// 1.024 MHz / 60 = 17066.6... cycles (23.2199 cycles per sample)
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// Need the last frame position in order to calculate correctly...
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// (or do we?)
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2007-05-29 03:06:33 +00:00
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2009-01-22 22:12:49 +00:00
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// SDL_LockAudio();
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SDL_mutexP(mutex2);
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2009-01-19 20:00:34 +00:00
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uint32 currentPos = sampleBase + (uint32)((double)time / CYCLES_PER_SAMPLE);
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2007-05-29 03:06:33 +00:00
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2009-01-19 20:00:34 +00:00
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if (currentPos > SOUND_BUFFER_SIZE - 1)
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2007-05-29 03:06:33 +00:00
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{
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#if 0
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WriteLog("ToggleSpeaker() about to go into spinlock at time: %08X (%u) (sampleBase=%u)!\n", time, time, sampleBase);
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#endif
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// Still hanging on this spinlock...
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// That could be because the "time" value is too high and so the buffer will NEVER be
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// empty enough...
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// Now that we're using a conditional, it seems to be working OK--though not perfectly...
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/*
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ToggleSpeaker() about to go into spinlock at time: 00004011 (16401) (sampleBase=3504)!
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16401 -> 706 samples, 3504 + 706 = 4210
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And it still thrashed the sound even though it didn't run into a spinlock...
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Seems like it's OK now that I've fixed the buffer-less-than-length bug...
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*/
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2009-01-22 22:12:49 +00:00
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// SDL_UnlockAudio();
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// SDL_CondWait(conditional, mutex);
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// SDL_LockAudio();
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currentPos = sampleBase + (uint32)((double)time / CYCLES_PER_SAMPLE);
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2007-05-29 03:06:33 +00:00
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#if 0
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WriteLog("--> after spinlock (sampleBase=%u)...\n", sampleBase);
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#endif
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}
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2009-01-19 16:09:37 +00:00
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int8 sample = (speakerState ? AMPLITUDE : -AMPLITUDE);
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2007-05-29 03:06:33 +00:00
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while (soundBufferPos < currentPos)
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2009-01-19 16:09:37 +00:00
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soundBuffer[soundBufferPos++] = (uint8)sample;
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2007-05-29 03:06:33 +00:00
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2009-01-22 22:12:49 +00:00
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// This is done *after* in case the buffer had a long dead spot (I think...)
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2007-05-29 03:06:33 +00:00
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speakerState = !speakerState;
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2009-01-22 22:12:49 +00:00
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SDL_mutexV(mutex2);
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// SDL_UnlockAudio();
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2007-05-29 03:06:33 +00:00
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}
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2009-01-22 22:12:49 +00:00
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void AddToSoundTimeBase(uint32 cycles)
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2007-05-29 03:06:33 +00:00
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{
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if (!soundInitialized)
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return;
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2009-01-22 22:12:49 +00:00
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// SDL_LockAudio();
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SDL_mutexP(mutex2);
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sampleBase += (uint32)((double)cycles / CYCLES_PER_SAMPLE);
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SDL_mutexV(mutex2);
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// SDL_UnlockAudio();
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2007-05-29 03:06:33 +00:00
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}
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2009-01-19 20:00:34 +00:00
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/*
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2009-01-22 22:12:49 +00:00
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HOW IT WORKS
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the main thread adds the amount of cpu time elapsed to samplebase. togglespeaker uses
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samplebase + current cpu time to find appropriate spot in buffer. it then fills the
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buffer up to the current time with the old toggle value before flipping it. the sound
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irq takes what it needs from the sound buffer and then adjusts both the buffer and
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samplebase back the appropriate amount.
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2009-01-19 20:00:34 +00:00
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A better way might be as follows:
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2009-01-22 22:12:49 +00:00
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Keep timestamp array of speaker toggle times. In the sound routine, unpack as many as will
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fit into the given buffer and keep going. Have the toggle function check to see if the
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buffer is full, and if it is, way for a signal from the interrupt that there's room for
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more. Can keep a circular buffer. Also, would need a timestamp buffer on the order of 2096
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samples *in theory* could toggle each sample
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2009-01-19 20:00:34 +00:00
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2009-01-22 22:12:49 +00:00
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Instead of a timestamp, just keep a delta. That way, don't need to deal with wrapping and
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all that (though the timestamp could wrap--need to check into that)
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2009-01-19 20:00:34 +00:00
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Need to consider corner cases where a sound IRQ happens but no speaker toggle happened.
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2009-01-22 22:12:49 +00:00
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If (delta > SAMPLES_PER_FRAME) then
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Here's the relevant cases:
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delta < SAMPLES_PER_FRAME -> Change happened within this time frame, so change buffer
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frame came and went, no change -> fill buffer with last value
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How to detect: Have bool bufferWasTouched = true when ToggleSpeaker() is called.
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Clear bufferWasTouched each frame.
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Two major cases here:
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o Buffer is touched on current frame
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o Buffer is untouched on current frame
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In the first case, it doesn't matter too much if the previous frame was touched or not,
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we don't really care except in finding the correct spot in the buffer to put our change
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in. In the second case, we need to tell the IRQ that nothing happened and to continue
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to output the same value.
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SO: How to synchronize the regular frame buffer with the IRQ buffer?
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What happens:
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Sound IRQ --> Every 1024 sample period (@ 44.1 KHz = 0.0232s)
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Emulation --> Render a frame --> 1/60 sec --> 735 samples
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--> sound buffer is filled
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Since the emulation is faster than the SIRQ the sound buffer should fill up
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prior to dumping it to the sound card.
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Problem is this: If silence happens for a long time then ToggleSpeaker is never
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called and the sound buffer has stale data; at least until soundBufferPos goes to
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zero and stays there...
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BUT this should be handled correctly by toggling the speaker value *after* filling
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the sound buffer...
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Still getting random clicks when running...
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(This may be due to the lock/unlock sound happening in ToggleSpeaker()...)
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2009-01-19 20:00:34 +00:00
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*/
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2009-01-22 22:12:49 +00:00
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