//
// Sound Interface
//
-// by James L. Hammons
-// (C) 2005 Underground Software
+// by James Hammons
+// (C) 2005-2018 Underground Software
//
-// JLH = James L. Hammons <jlhamm@acm.org>
+// JLH = James Hammons <jlhamm@acm.org>
//
// WHO WHEN WHAT
-// --- ---------- ------------------------------------------------------------
+// --- ---------- -----------------------------------------------------------
// JLH 12/02/2005 Fixed a problem with sound callback thread signaling the
// main thread
-// JLH 12/03/2005 Fixed sound callback dropping samples when the sample buffer
-// is shorter than the callback sample buffer
+// JLH 12/03/2005 Fixed sound callback dropping samples when the sample
+// buffer is shorter than the callback sample buffer
//
// 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 <string.h> // For memset, memcpy
-#include <SDL.h>
+#include <string.h> // For memset, memcpy
+#include <SDL2/SDL.h>
#include "log.h"
+#include "mockingboard.h"
-//#define SAMPLE_RATE (44100.0)
-#define SAMPLE_RATE (48000.0)
+// Useful defines
+
+//#define DEBUG
+
#define SAMPLES_PER_FRAME (SAMPLE_RATE / 60.0)
#define CYCLES_PER_SAMPLE (1024000.0 / SAMPLE_RATE)
-#define SOUND_BUFFER_SIZE (8192)
-//#define AMPLITUDE (16) // -32 - +32 seems to be plenty loud!
+// 32K ought to be enough for anybody
+#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 bool speakerState = false;
-static uint8 soundBuffer[SOUND_BUFFER_SIZE];
-static uint32 soundBufferPos;
-static uint32 sampleBase;
-static uint64 lastToggleCycles;
-static uint64 samplePosition;
-static SDL_cond * conditional = NULL;
-static SDL_mutex * mutex = NULL;
-static SDL_mutex * mutex2 = NULL;
-static uint8 ampPtr = 5;
-static uint16 amplitude[17] = { 0, 1, 2, 4, 8, 16, 32, 64, 128, 256, 512, 1024, 2048,
- 4096, 8192, 16384, 32768 };
+static uint16_t soundBuffer[SOUND_BUFFER_SIZE];
+static uint32_t soundBufferPos;
+static uint16_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 };
// Private function prototypes
static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length);
+
+/*
+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. :-/
+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!).
+*/
//
// Initialize the SDL sound system
//
void SoundInit(void)
{
-#if 0
-// To weed out problems for now...
-return;
-#endif
-
- 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)...
+ 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_U16SYS; // 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
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");
+ WriteLog("SDL sez: %s\n", SDL_GetError());
return;
}
- conditional = SDL_CreateCond();
- mutex = SDL_CreateMutex();
- mutex2 = SDL_CreateMutex();// Let's try real signalling...
- SDL_mutexP(mutex); // Must lock the mutex for the cond to work properly...
soundBufferPos = 0;
- sampleBase = 0;
- lastToggleCycles = 0;
- samplePosition = 0;
+ sample = desired.silence; // ? wilwok ? yes
- SDL_PauseAudio(false); // Start playback!
+ SDL_PauseAudioDevice(device, 0);// Start playback!
soundInitialized = true;
WriteLog("Sound: Successfully initialized.\n");
}
+
//
// Close down the SDL sound subsystem
//
{
if (soundInitialized)
{
- SDL_PauseAudio(true);
- SDL_CloseAudio();
- SDL_DestroyCond(conditional);
- SDL_DestroyMutex(mutex);
- SDL_DestroyMutex(mutex2);
+ SDL_PauseAudioDevice(device, 1);
+ SDL_CloseAudioDevice(device);
WriteLog("Sound: Done.\n");
}
}
+
+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 uint32_t sndFrmCnt = 0;
+static uint32_t lastStarve = 0;
+static void SDLSoundCallback(void * /*userdata*/, Uint8 * buffer8, int length8)
{
- // 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.)
- // (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!)
+sndFrmCnt++;
- // Let's try using a mutex for shared resource consumption...
- SDL_mutexP(mutex2);
+ // Recast this as a 16-bit type...
+ uint16_t * buffer = (uint16_t *)buffer8;
+ uint32_t length = (uint32_t)length8 / 2;
- if (soundBufferPos < (uint32)length) // The sound buffer is starved...
+ if (soundBufferPos < length)
{
-//printf("Sound buffer starved!\n");
-//fflush(stdout);
- for(uint32 i=0; i<soundBufferPos; i++)
+//WriteLog("*** Sound buffer starved (%d short) *** [%d delta %d]\n", length - soundBufferPos, sndFrmCnt, sndFrmCnt - lastStarve);
+lastStarve = sndFrmCnt;
+#if 1
+ for(uint32_t i=0; i<length; i++)
+ buffer[i] = desired.silence;
+#else
+ // 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)(speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]), length - soundBufferPos);
- soundBufferPos = 0; // Reset soundBufferPos to start of buffer...
- sampleBase = 0; // & sampleBase...
-//Ick. This should never happen!
-//Actually, this probably happens a lot. (?)
-// SDL_CondSignal(conditional); // Wake up any threads waiting for the buffer to drain...
-// return; // & bail!
+ for(uint32_t i=soundBufferPos; i<length; i++)
+ buffer[i] = sample;
+
+ // Reset soundBufferPos to start of buffer...
+ soundBufferPos = 0;
+#endif
}
else
{
// Fill sound buffer with frame buffered sound
- memcpy(buffer, soundBuffer, length);
+ for(uint32_t i=0; i<length; i++)
+ buffer[i] = soundBuffer[i];
+
soundBufferPos -= length;
- sampleBase -= 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];
}
-
- // Update our sample position
- samplePosition += length;
- // Free the mutex...
- SDL_mutexV(mutex2);
- // Wake up any threads waiting for the buffer to drain...
- SDL_CondSignal(conditional);
}
-// 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 ToggleSpeaker(uint64 elapsedCycles)
-{
- if (!soundInitialized)
- return;
- uint64 deltaCycles = elapsedCycles - lastToggleCycles;
-
-#if 0
-if (time > 95085)//(time & 0x80000000)
+//
+// This is called by the main CPU thread every ~21.666 cycles.
+//
+void WriteSampleToBuffer(void)
{
- WriteLog("ToggleSpeaker() given bad time value: %08X (%u)!\n", time, time);
-// fflush(stdout);
-}
-#endif
-
- // 1.024 MHz / 60 = 17066.6... cycles (23.2199 cycles per sample)
- // Need the last frame position in order to calculate correctly...
- // (or do we?)
-
-// SDL_LockAudio();
- SDL_mutexP(mutex2);
-// uint32 currentPos = sampleBase + (uint32)((double)elapsedCycles / CYCLES_PER_SAMPLE);
- uint32 currentPos = (uint32)((double)deltaCycles / CYCLES_PER_SAMPLE);
-
-/*
-The problem:
-
- ______ | ______________ | ______
-____| | | |_______
-
-Speaker is toggled, then not toggled for a while. How to find buffer position in the
-last frame?
+// uint16_t s1 = AYGetSample(0);
+// uint16_t s2 = AYGetSample(1);
+ uint16_t s1 = mb[0].ay[0].GetSample();
+ uint16_t s2 = mb[0].ay[1].GetSample();
-IRQ buffer len is 1024.
-
-Could check current CPU clock, take delta. If delta > 1024, then ...
-
-Could add # of cycles in IRQ to lastToggleCycles, then currentPos will be guaranteed
-to fall within acceptable limits.
-*/
-
-
- if (currentPos > SOUND_BUFFER_SIZE - 1)
+ // This should almost never happen, but, if it does...
+ while (soundBufferPos >= (SOUND_BUFFER_SIZE - 1))
{
-#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);
-// SDL_LockAudio();
-// Hm.
-SDL_mutexV(mutex2);//Release it so sound thread can get it,
-SDL_CondWait(conditional, mutex);//Sleep/wait for the sound thread
-SDL_mutexP(mutex2);//Re-lock it until we're done with it...
-
- currentPos = sampleBase + (uint32)((double)elapsedCycles / CYCLES_PER_SAMPLE);
-#if 0
-WriteLog("--> after spinlock (sampleBase=%u)...\n", sampleBase);
-#endif
+//WriteLog("WriteSampleToBuffer(): Waiting for sound thread. soundBufferPos=%i, SOUNDBUFFERSIZE-1=%i\n", soundBufferPos, SOUND_BUFFER_SIZE-1);
+ SDL_Delay(1);
}
- int8 sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]);
-
- while (soundBufferPos < currentPos)
- soundBuffer[soundBufferPos++] = (uint8)sample;
-
- // This is done *after* in case the buffer had a long dead spot (I think...)
- speakerState = !speakerState;
- SDL_mutexV(mutex2);
-// SDL_UnlockAudio();
+ SDL_LockAudioDevice(device);
+ soundBuffer[soundBufferPos++] = sample + s1 + s2;
+ SDL_UnlockAudioDevice(device);
}
-void AddToSoundTimeBase(uint32 cycles)
+
+void ToggleSpeaker(void)
{
if (!soundInitialized)
return;
-// SDL_LockAudio();
- SDL_mutexP(mutex2);
- sampleBase += (uint32)((double)cycles / CYCLES_PER_SAMPLE);
- SDL_mutexV(mutex2);
-// SDL_UnlockAudio();
+ speakerState = !speakerState;
+ sample = (speakerState ? amplitude[ampPtr] : 0);
}
+
void VolumeUp(void)
{
- // Currently set for 8-bit samples
- if (ampPtr < 8)
+ // 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