415 lines
13 KiB
C++
Executable File
415 lines
13 KiB
C++
Executable File
//
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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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// - Figure out why it's playing too fast
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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 <SDL2/SDL.h>
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#include "log.h"
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// Useful defines
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//#define DEBUG
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//#define WRITE_OUT_WAVE
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// This is odd--seems to be working properly now! Maybe a bug in the SDL sound code?
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// Actually, it still doesn't sound right... Sounds too slow now. :-/
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// But then again, it's difficult to tell. Sometimes it slows waaaaaay down, but generally
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// seems to be OK other than that
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// Also, it could be that the discrepancy in pitch is due to the V65C02 and it's lack of
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// cycle accuracy...
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//#define SAMPLE_RATE (44100.0)
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#define SAMPLE_RATE (48000.0)
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#define SAMPLES_PER_FRAME (SAMPLE_RATE / 60.0)
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// This works for AppleWin but not here... ??? WHY ???
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// ~ 21
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#define CYCLES_PER_SAMPLE (1024000.0 / SAMPLE_RATE)
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// ~ 17 (lower pitched than above...!)
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// Makes sense, as this is the divisor for # of cycles passed
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//#define CYCLES_PER_SAMPLE (800000.0 / SAMPLE_RATE)
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// This seems about right, compared to AppleWin--but AW runs @ 1.024 MHz
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// 23 (1.024) vs. 20 (0.900)
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//#define CYCLES_PER_SAMPLE (900000.0 / SAMPLE_RATE)
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//nope, too high #define CYCLES_PER_SAMPLE (960000.0 / SAMPLE_RATE)
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//#define CYCLES_PER_SAMPLE 21
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//#define SOUND_BUFFER_SIZE (8192)
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#define SOUND_BUFFER_SIZE (32768)
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// Global variables
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// Local variables
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static SDL_AudioSpec desired, obtained;
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static SDL_AudioDeviceID device;
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static bool soundInitialized = false;
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static bool speakerState = false;
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static int16_t soundBuffer[SOUND_BUFFER_SIZE];
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static uint32_t soundBufferPos;
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static uint64_t lastToggleCycles;
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static SDL_cond * conditional = NULL;
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static SDL_mutex * mutex = NULL;
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static SDL_mutex * mutex2 = NULL;
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static int16_t sample;
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static uint8_t ampPtr = 12; // Start with -2047 - +2047
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static int16_t amplitude[17] = { 0, 1, 2, 3, 7, 15, 31, 63, 127, 255, 511, 1023, 2047,
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4095, 8191, 16383, 32767 };
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#ifdef WRITE_OUT_WAVE
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static FILE * fp = NULL;
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#endif
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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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#if 0
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// To weed out problems for now...
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return;
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#endif
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SDL_zero(desired);
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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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desired.format = AUDIO_S16SYS; // This uses the native endian (for portability)...
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desired.channels = 1;
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desired.samples = 512; // Let's try a 1/2K buffer (can always go lower)
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desired.callback = SDLSoundCallback;
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device = SDL_OpenAudioDevice(NULL, 0, &desired, &obtained, 0);
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if (device == 0)
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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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mutex2 = SDL_CreateMutex();// Let's try real signalling...
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soundBufferPos = 0;
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lastToggleCycles = 0;
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sample = desired.silence; // ? wilwok ? yes
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SDL_PauseAudioDevice(device, 0); // Start playback!
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soundInitialized = true;
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WriteLog("Sound: Successfully initialized.\n");
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#ifdef WRITE_OUT_WAVE
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fp = fopen("./apple2.wav", "wb");
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#endif
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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_PauseAudioDevice(device, 1);
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// SDL_CloseAudio();
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SDL_CloseAudioDevice(device);
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SDL_DestroyCond(conditional);
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SDL_DestroyMutex(mutex);
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SDL_DestroyMutex(mutex2);
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WriteLog("Sound: Done.\n");
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#ifdef WRITE_OUT_WAVE
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fclose(fp);
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#endif
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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 * buffer8, int length8)
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{
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//WriteLog("SDLSoundCallback(): begin (soundBufferPos=%i)\n", soundBufferPos);
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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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// (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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// (Should NOT starve now, now that we properly handle frame edges...)
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// Let's try using a mutex for shared resource consumption...
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//Actually, I think Lock/UnlockAudio() does this already...
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//WriteLog("SDLSoundCallback: soundBufferPos = %i\n", soundBufferPos);
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SDL_mutexP(mutex2);
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// Recast this as a 16-bit type...
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int16_t * buffer = (int16_t *)buffer8;
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uint32_t length = (uint32_t)length8 / 2;
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//WriteLog("SDLSoundCallback(): filling buffer...\n");
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if (soundBufferPos < length) // The sound buffer is starved...
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{
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for(uint32_t i=0; i<soundBufferPos; i++)
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buffer[i] = soundBuffer[i];
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// Fill buffer with last value
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// memset(buffer + soundBufferPos, (uint8_t)sample, length - soundBufferPos);
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for(uint32_t i=soundBufferPos; i<length; i++)
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buffer[i] = sample;
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soundBufferPos = 0; // Reset soundBufferPos to start of buffer...
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}
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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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for(uint32_t i=0; i<length; i++)
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buffer[i] = soundBuffer[i];
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soundBufferPos -= length;
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// Move current buffer down to start
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for(uint32_t i=0; i<soundBufferPos; i++)
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soundBuffer[i] = soundBuffer[length + i];
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}
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// Free the mutex...
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//WriteLog("SDLSoundCallback(): SDL_mutexV(mutex2)\n");
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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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//WriteLog("SDLSoundCallback(): end\n");
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}
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// This is called by the main CPU thread every ~21.333 cycles.
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void WriteSampleToBuffer(void)
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{
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//WriteLog("WriteSampleToBuffer(): SDL_mutexP(mutex2)\n");
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SDL_mutexP(mutex2);
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// This should almost never happen, but...
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while (soundBufferPos >= (SOUND_BUFFER_SIZE - 1))
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{
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//WriteLog("WriteSampleToBuffer(): Waiting for sound thread. soundBufferPos=%i, SOUNDBUFFERSIZE-1=%i\n", soundBufferPos, SOUND_BUFFER_SIZE-1);
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SDL_mutexV(mutex2); // Release it so sound thread can get it,
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SDL_mutexP(mutex); // Must lock the mutex for the cond to work properly...
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SDL_CondWait(conditional, mutex); // Sleep/wait for the sound thread
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SDL_mutexV(mutex); // Must unlock the mutex for the cond to work properly...
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SDL_mutexP(mutex2); // Re-lock it until we're done with it...
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}
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soundBuffer[soundBufferPos++] = sample;
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//WriteLog("WriteSampleToBuffer(): SDL_mutexV(mutex2)\n");
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SDL_mutexV(mutex2);
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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 HandleBuffer(uint64_t elapsedCycles)
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{
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// Step 1: Calculate delta time
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uint64_t deltaCycles = elapsedCycles - lastToggleCycles;
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// Step 2: Calculate new buffer position
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uint32_t currentPos = (uint32_t)((double)deltaCycles / CYCLES_PER_SAMPLE);
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// Step 3: Make sure there's room for it
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// We need to lock since we touch both soundBuffer and soundBufferPos
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SDL_mutexP(mutex2);
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while ((soundBufferPos + currentPos) > (SOUND_BUFFER_SIZE - 1))
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{
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SDL_mutexV(mutex2); // Release it so sound thread can get it,
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SDL_mutexP(mutex); // Must lock the mutex for the cond to work properly...
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SDL_CondWait(conditional, mutex); // Sleep/wait for the sound thread
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SDL_mutexV(mutex); // Must unlock the mutex for the cond to work properly...
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SDL_mutexP(mutex2); // Re-lock it until we're done with it...
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}
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// Step 4: Backfill and adjust lastToggleCycles
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// currentPos is position from "zero" or soundBufferPos...
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currentPos += soundBufferPos;
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#ifdef WRITE_OUT_WAVE
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uint32_t sbpSave = soundBufferPos;
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#endif
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// Backfill with current toggle state
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while (soundBufferPos < currentPos)
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soundBuffer[soundBufferPos++] = sample;
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#ifdef WRITE_OUT_WAVE
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fwrite(&soundBuffer[sbpSave], sizeof(int16_t), currentPos - sbpSave, fp);
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#endif
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SDL_mutexV(mutex2);
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lastToggleCycles = elapsedCycles;
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}
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void ToggleSpeaker(uint64_t elapsedCycles)
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{
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if (!soundInitialized)
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return;
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// HandleBuffer(elapsedCycles);
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speakerState = !speakerState;
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sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]);
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}
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void AdjustLastToggleCycles(uint64_t elapsedCycles)
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{
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if (!soundInitialized)
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return;
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/*
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BOOKKEEPING
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We need to know the following:
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o Where in the sound buffer the base or "zero" time is
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o At what CPU timestamp the speaker was last toggled
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NOTE: we keep things "right" by advancing this number every frame, even
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if nothing happened! That way, we can keep track without having
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to detect whether or not several frames have gone by without any
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activity.
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How to do it:
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Every time the speaker is toggled, we move the base or "zero" time to the
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current spot in the buffer. We also backfill the buffer up to that point with
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the old toggle value. The next time the speaker is toggled, we measure the
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difference in time between the last time it was toggled (the "zero") and now,
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and repeat the cycle.
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We handle dead spots by backfilling the buffer with the current toggle value
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every frame--this way we don't have to worry about keeping current time and
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crap like that. So, we have to move the "zero" the right amount, just like
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in ToggleSpeaker(), and backfill only without toggling.
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*/
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HandleBuffer(elapsedCycles);
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}
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void VolumeUp(void)
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{
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// Currently set for 16-bit samples
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if (ampPtr < 16)
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ampPtr++;
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}
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void VolumeDown(void)
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{
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if (ampPtr > 0)
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ampPtr--;
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}
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uint8_t GetVolume(void)
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{
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return ampPtr;
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}
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/*
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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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A better way might be as follows:
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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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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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Need to consider corner cases where a sound IRQ happens but no speaker toggle happened.
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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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*/
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