//
// Sound Interface
//
-// by James L. Hammons
+// by James Hammons
// (C) 2005 Underground Software
//
-// JLH = James L. Hammons <jlhamm@acm.org>
+// JLH = James Hammons <jlhamm@acm.org>
//
// WHO WHEN WHAT
// --- ---------- ------------------------------------------------------------
// STILL TO DO:
//
// - Figure out why it's losing samples (Bard's Tale) [DONE]
-// - Figure out why it's playing too fast
+// - Figure out why it's playing too fast [DONE]
//
#include "sound.h"
#include <string.h> // For memset, memcpy
-#include <SDL.h>
+#include <SDL2/SDL.h>
#include "log.h"
// Useful defines
//#define DEBUG
//#define WRITE_OUT_WAVE
-// This is odd--seems to be working properly now! Maybe a bug in the SDL sound code?
-// Actually, it still doesn't sound right... Sounds too slow now. :-/
-// But then again, it's difficult to tell. Sometimes it slows waaaaaay down, but generally
-// seems to be OK other than that
-// Also, it could be that the discrepancy in pitch is due to the V65C02 and it's lack of
-// cycle accuracy...
-
//#define SAMPLE_RATE (44100.0)
#define SAMPLE_RATE (48000.0)
#define SAMPLES_PER_FRAME (SAMPLE_RATE / 60.0)
-// This works for AppleWin but not here... ??? WHY ???
-// ~ 21
#define CYCLES_PER_SAMPLE (1024000.0 / SAMPLE_RATE)
-// ~ 17 (lower pitched than above...!)
-// Makes sense, as this is the divisor for # of cycles passed
-//#define CYCLES_PER_SAMPLE (800000.0 / SAMPLE_RATE)
-// This seems about right, compared to AppleWin--but AW runs @ 1.024 MHz
-// 23 (1.024) vs. 20 (0.900)
-//#define CYCLES_PER_SAMPLE (900000.0 / SAMPLE_RATE)
-//nope, too high #define CYCLES_PER_SAMPLE (960000.0 / SAMPLE_RATE)
-//#define CYCLES_PER_SAMPLE 21
//#define SOUND_BUFFER_SIZE (8192)
#define SOUND_BUFFER_SIZE (32768)
// Local variables
static SDL_AudioSpec desired, obtained;
+static SDL_AudioDeviceID device;
static bool soundInitialized = false;
static bool speakerState = false;
-static int16 soundBuffer[SOUND_BUFFER_SIZE];
-static uint32 soundBufferPos;
-static uint64 lastToggleCycles;
+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 sample;
-static uint8 ampPtr = 14; // Start with -16 - +16
-static int16 amplitude[17] = { 0, 1, 2, 3, 7, 15, 31, 63, 127, 255, 511, 1023, 2047,
+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;
static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length);
+
//
// Initialize the SDL sound system
//
// 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_S8; // This uses the native endian (for portability)...
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 = 1024; // Let's try a 1K 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
-//When doing it this way, we need to check to see if we got what we asked for...
- if (SDL_OpenAudio(&desired, &obtained) < 0)
+ device = SDL_OpenAudioDevice(NULL, 0, &desired, &obtained, 0);
+
+ if (device == 0)
{
WriteLog("Sound: Failed to initialize SDL sound.\n");
return;
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");
#endif
}
+
//
// Close down the SDL sound subsystem
//
{
if (soundInitialized)
{
- SDL_PauseAudio(true);
- SDL_CloseAudio();
+ SDL_PauseAudioDevice(device, 1);
+ SDL_CloseAudioDevice(device);
SDL_DestroyCond(conditional);
SDL_DestroyMutex(mutex);
SDL_DestroyMutex(mutex2);
}
}
+
+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)
+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.)
// 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 * buffer = (int16 *)buffer8;
- uint32 length = (uint32)length8 / 2;
+ int16_t * buffer = (int16_t *)buffer8;
+ uint32_t length = (uint32_t)length8 / 2;
- if (soundBufferPos < length) // The sound buffer is starved...
+//WriteLog("SDLSoundCallback(): filling buffer...\n");
+ if (soundBufferPos < length)
{
- for(uint32 i=0; i<soundBufferPos; i++)
+ // The sound buffer is starved...
+ for(uint32_t i=0; i<soundBufferPos; i++)
buffer[i] = soundBuffer[i];
// Fill buffer with last value
-// memset(buffer + soundBufferPos, (uint8)sample, length - soundBufferPos);
- for(uint32 i=soundBufferPos; i<length; i++)
- buffer[i] = (uint16)sample;
- soundBufferPos = 0; // Reset soundBufferPos to start of buffer...
+ 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
-// memcpy(buffer, soundBuffer, length);
- for(uint32 i=0; i<length; i++)
+ for(uint32_t i=0; i<length; i++)
buffer[i] = soundBuffer[i];
+
soundBufferPos -= length;
// Move current buffer down to start
- for(uint32 i=0; i<soundBufferPos; i++)
+ 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");
}
-// Need some interface functions here to take care of flipping the
-// waveform at the correct time in the sound stream...
-
-/*
-Maybe set up a buffer 1 frame long (44100 / 60 = 735 bytes per frame)
-
-Hmm. That's smaller than the sound buffer 2048 bytes... (About 2.75 frames needed to fill)
-
-So... I guess what we could do is this:
-
--- 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 HandleBuffer(uint64 elapsedCycles)
+// This is called by the main CPU thread every ~21.333 cycles.
+void WriteSampleToBuffer(void)
{
- // Step 1: Calculate delta time
- uint64 deltaCycles = elapsedCycles - lastToggleCycles;
-
- // Step 2: Calculate new buffer position
- uint32 currentPos = (uint32)((double)deltaCycles / CYCLES_PER_SAMPLE);
-
- // Step 3: Make sure there's room for it
- // We need to lock since we touch both soundBuffer and soundBufferPos
+//WriteLog("WriteSampleToBuffer(): SDL_mutexP(mutex2)\n");
SDL_mutexP(mutex2);
- while ((soundBufferPos + currentPos) > (SOUND_BUFFER_SIZE - 1))
+
+ // This should almost never happen, but, if it does...
+ while (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...
+//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...
}
- // Step 4: Backfill and adjust lastToggleCycles
- // currentPos is position from "zero" or soundBufferPos...
- currentPos += soundBufferPos;
-
-#ifdef WRITE_OUT_WAVE
- uint32 sbpSave = soundBufferPos;
-#endif
- // Backfill with current toggle state
- while (soundBufferPos < currentPos)
- soundBuffer[soundBufferPos++] = (uint16)sample;
-
-#ifdef WRITE_OUT_WAVE
- fwrite(&soundBuffer[sbpSave], sizeof(int16), currentPos - sbpSave, fp);
-#endif
-
+ soundBuffer[soundBufferPos++] = sample;
+//WriteLog("WriteSampleToBuffer(): SDL_mutexV(mutex2)\n");
SDL_mutexV(mutex2);
- lastToggleCycles = elapsedCycles;
}
-void ToggleSpeaker(uint64 elapsedCycles)
+
+void ToggleSpeaker(void)
{
if (!soundInitialized)
return;
- HandleBuffer(elapsedCycles);
speakerState = !speakerState;
sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]);
}
-void AdjustLastToggleCycles(uint64 elapsedCycles)
-{
- if (!soundInitialized)
- return;
-/*
-BOOKKEEPING
-
-We need to know the following:
-
- o Where in the sound buffer the base or "zero" time is
- o At what CPU timestamp the speaker was last toggled
- NOTE: we keep things "right" by advancing this number every frame, even
- if nothing happened! That way, we can keep track without having
- to detect whether or not several frames have gone by without any
- activity.
-
-How to do it:
-
-Every time the speaker is toggled, we move the base or "zero" time to the
-current spot in the buffer. We also backfill the buffer up to that point with
-the old toggle value. The next time the speaker is toggled, we measure the
-difference in time between the last time it was toggled (the "zero") and now,
-and repeat the cycle.
-
-We handle dead spots by backfilling the buffer with the current toggle value
-every frame--this way we don't have to worry about keeping current time and
-crap like that. So, we have to move the "zero" the right amount, just like
-in ToggleSpeaker(), and backfill only without toggling.
-*/
- HandleBuffer(elapsedCycles);
-}
void VolumeUp(void)
{
- // Currently set for 8-bit samples
- // Now 16
+ // Currently set for 16-bit samples
if (ampPtr < 16)
ampPtr++;
}
+
void VolumeDown(void)
{
if (ampPtr > 0)
ampPtr--;
}
-uint8 GetVolume(void)
+
+uint8_t GetVolume(void)
{
return ampPtr;
}
-/*
-HOW IT WORKS
-
-the main thread adds the amount of cpu time elapsed to samplebase. togglespeaker uses
-samplebase + current cpu time to find appropriate spot in buffer. it then fills the
-buffer up to the current time with the old toggle value before flipping it. the sound
-irq takes what it needs from the sound buffer and then adjusts both the buffer and
-samplebase back the appropriate amount.
-
-
-A better way might be as follows:
-
-Keep timestamp array of speaker toggle times. In the sound routine, unpack as many as will
-fit into the given buffer and keep going. Have the toggle function check to see if the
-buffer is full, and if it is, way for a signal from the interrupt that there's room for
-more. Can keep a circular buffer. Also, would need a timestamp buffer on the order of 2096
-samples *in theory* could toggle each sample
-
-Instead of a timestamp, just keep a delta. That way, don't need to deal with wrapping and
-all that (though the timestamp could wrap--need to check into that)
-
-Need to consider corner cases where a sound IRQ happens but no speaker toggle happened.
-
-If (delta > SAMPLES_PER_FRAME) then
-
-Here's the relevant cases:
-
-delta < SAMPLES_PER_FRAME -> Change happened within this time frame, so change buffer
-frame came and went, no change -> fill buffer with last value
-How to detect: Have bool bufferWasTouched = true when ToggleSpeaker() is called.
-Clear bufferWasTouched each frame.
-
-Two major cases here:
-
- o Buffer is touched on current frame
- o Buffer is untouched on current frame
-
-In the first case, it doesn't matter too much if the previous frame was touched or not,
-we don't really care except in finding the correct spot in the buffer to put our change
-in. In the second case, we need to tell the IRQ that nothing happened and to continue
-to output the same value.
-
-SO: How to synchronize the regular frame buffer with the IRQ buffer?
-
-What happens:
- Sound IRQ --> Every 1024 sample period (@ 44.1 KHz = 0.0232s)
- Emulation --> Render a frame --> 1/60 sec --> 735 samples
- --> sound buffer is filled
-
-Since the emulation is faster than the SIRQ the sound buffer should fill up
-prior to dumping it to the sound card.
-
-Problem is this: If silence happens for a long time then ToggleSpeaker is never
-called and the sound buffer has stale data; at least until soundBufferPos goes to
-zero and stays there...
-
-BUT this should be handled correctly by toggling the speaker value *after* filling
-the sound buffer...
-
-Still getting random clicks when running...
-(This may be due to the lock/unlock sound happening in ToggleSpeaker()...)
-*/
-
projects / apple2 / blobdiff