224 lines
5.6 KiB
C++
224 lines
5.6 KiB
C++
//
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// Sound Interface
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//
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// by James Hammons
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// (C) 2005-2018 Underground Software
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//
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// JLH = James 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
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// buffer 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 [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 <SDL2/SDL.h>
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#include "log.h"
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#include "mockingboard.h"
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// Useful defines
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//#define DEBUG
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#define SAMPLES_PER_FRAME (SAMPLE_RATE / 60.0)
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#define CYCLES_PER_SAMPLE (1024000.0 / SAMPLE_RATE)
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// 32K ought to be enough for anybody
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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 uint16_t soundBuffer[SOUND_BUFFER_SIZE];
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static uint32_t soundBufferPos;
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static uint16_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,
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511, 1023, 2047, 4095, 8191, 16383, 32767 };
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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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N.B: We can convert this from the current callback model to a push model by using SDL_QueueAudio(SDL_AudioDeviceID id, const void * data, Uint32 len) where id is the audio device ID, data is a pointer to the sound buffer, and len is the size of the buffer in *bytes* (not samples!). To use this method, we need to set up things as usual but instead of putting the callback function pointer in desired.callback, we put a NULL there. The downside is that we can't tell if the buffer is being starved or not, which is why we haven't kicked it to the curb just yet--we want to know why we're still getting buffer starvation even if it's not as frequent as it used to be. :-/
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You can get the size of the audio already queued with SDL_GetQueuedAudioSize(SDL_AudioDeviceID id), which will return the size of the buffer in bytes (again, *not* samples!).
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*/
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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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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_U16SYS; // 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
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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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WriteLog("SDL sez: %s\n", SDL_GetError());
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return;
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}
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soundBufferPos = 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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}
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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_PauseAudioDevice(device, 1);
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SDL_CloseAudioDevice(device);
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WriteLog("Sound: Done.\n");
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}
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}
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void SoundPause(void)
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{
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if (soundInitialized)
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SDL_PauseAudioDevice(device, 1);
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}
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void SoundResume(void)
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{
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if (soundInitialized)
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SDL_PauseAudioDevice(device, 0);
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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 uint32_t sndFrmCnt = 0;
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static uint32_t lastStarve = 0;
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static void SDLSoundCallback(void * /*userdata*/, Uint8 * buffer8, int length8)
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{
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sndFrmCnt++;
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// Recast this as a 16-bit type...
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uint16_t * buffer = (uint16_t *)buffer8;
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uint32_t length = (uint32_t)length8 / 2;
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if (soundBufferPos < length)
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{
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//WriteLog("*** Sound buffer starved (%d short) *** [%d delta %d]\n", length - soundBufferPos, sndFrmCnt, sndFrmCnt - lastStarve);
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lastStarve = sndFrmCnt;
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#if 1
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for(uint32_t i=0; i<length; i++)
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buffer[i] = desired.silence;
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#else
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// The sound buffer is starved...
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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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for(uint32_t i=soundBufferPos; i<length; i++)
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buffer[i] = sample;
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// Reset soundBufferPos to start of buffer...
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soundBufferPos = 0;
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#endif
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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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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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}
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//
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// This is called by the main CPU thread every ~21.666 cycles.
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//
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void WriteSampleToBuffer(void)
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{
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// uint16_t s1 = AYGetSample(0);
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// uint16_t s2 = AYGetSample(1);
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uint16_t s1 = mb[0].ay[0].GetSample();
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uint16_t s2 = mb[0].ay[1].GetSample();
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// This should almost never happen, but, if it does...
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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_Delay(1);
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}
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SDL_LockAudioDevice(device);
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soundBuffer[soundBufferPos++] = sample + s1 + s2;
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SDL_UnlockAudioDevice(device);
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}
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void ToggleSpeaker(void)
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{
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if (!soundInitialized)
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return;
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speakerState = !speakerState;
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sample = (speakerState ? amplitude[ampPtr] : 0);
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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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