X-Git-Url: http://shamusworld.gotdns.org/cgi-bin/gitweb.cgi?a=blobdiff_plain;f=src%2Fsound.cpp;h=75f9ba2bae9779c3493752f28867905f59fd0d8d;hb=f36d026c7b8b398b88765ec5b67a3c767fe5fbad;hp=53c6fb05ef6389df9fb366bb6faecaff51675e65;hpb=d0e4cf72abe47561d7e2c371b104ea77a4b6a086;p=apple2 diff --git a/src/sound.cpp b/src/sound.cpp old mode 100755 new mode 100644 index 53c6fb0..75f9ba2 --- a/src/sound.cpp +++ b/src/sound.cpp @@ -1,10 +1,10 @@ // // Sound Interface // -// by James L. Hammons +// by James Hammons // (C) 2005 Underground Software // -// JLH = James L. Hammons +// JLH = James Hammons // // WHO WHEN WHAT // --- ---------- ------------------------------------------------------------ @@ -17,56 +17,55 @@ // 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 // For memset, memcpy -#include +#include #include "log.h" // Useful defines //#define DEBUG +//#define WRITE_OUT_WAVE //#define SAMPLE_RATE (44100.0) #define SAMPLE_RATE (48000.0) #define SAMPLES_PER_FRAME (SAMPLE_RATE / 60.0) -// ~ 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) -//nope, too high #define CYCLES_PER_SAMPLE (960000.0 / SAMPLE_RATE) +#define CYCLES_PER_SAMPLE (1024000.0 / SAMPLE_RATE) //#define SOUND_BUFFER_SIZE (8192) -#define SOUND_BUFFER_SIZE (16384) +#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 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 int8 sample; -static uint8 ampPtr = 5; // Start with -16 - +16 -static uint16 amplitude[17] = { 0, 1, 2, 4, 8, 16, 32, 64, 128, 256, 512, 1024, 2048, - 4096, 8192, 16384, 32768 }; +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; +#endif // Private function prototypes static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length); + // // Initialize the SDL sound system // @@ -76,17 +75,16 @@ void SoundInit(void) // 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.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 + device = SDL_OpenAudioDevice(NULL, 0, &desired, &obtained, 0); + + if (device == 0) { WriteLog("Sound: Failed to initialize SDL sound.\n"); return; @@ -95,18 +93,20 @@ 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"); + +#ifdef WRITE_OUT_WAVE + fp = fopen("./apple2.wav", "wb"); +#endif } + // // Close down the SDL sound subsystem // @@ -114,20 +114,40 @@ void SoundDone(void) { if (soundInitialized) { - SDL_PauseAudio(true); - SDL_CloseAudio(); + SDL_PauseAudioDevice(device, 1); + SDL_CloseAudioDevice(device); SDL_DestroyCond(conditional); SDL_DestroyMutex(mutex); SDL_DestroyMutex(mutex2); WriteLog("Sound: Done.\n"); + +#ifdef WRITE_OUT_WAVE + fclose(fp); +#endif } } + +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 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.) @@ -136,340 +156,100 @@ static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length) // (Should NOT starve now, now that we properly handle frame edges...) // 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); - if (soundBufferPos < (uint32)length) // The sound buffer is starved... + // Recast this as a 16-bit type... + int16_t * buffer = (int16_t *)buffer8; + uint32_t length = (uint32_t)length8 / 2; + +//WriteLog("SDLSoundCallback(): filling buffer...\n"); + if (soundBufferPos < length) { -//printf("Sound buffer starved!\n"); -//fflush(stdout); - for(uint32 i=0; i 95085)//(time & 0x80000000) -{ - 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(); +//WriteLog("WriteSampleToBuffer(): SDL_mutexP(mutex2)\n"); 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? - -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. - -This *should* work, but if the IRQ isn't scheduled & etc, could screw timing up. -Need to have a way to suspend IRQ thread as well as CPU thread when in the GUI, -for example - -Another method would be to add to lastToggleCycles on every timeslice of the CPU, -just like we used to. - -Or, run the CPU for CYCLES_PER_SAMPLE and take a sample, then copy the buffer over -at the end of the timeslice. That way, we could just fill the buffer and let the -IRQ handle draining it. No muss, no fuss. -*/ - - - if ((soundBufferPos + 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. -// This might not empty the buffer enough, causing hash and trash. !!! FIX !!! -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)deltaCycles / CYCLES_PER_SAMPLE); - currentPos = (uint32)((double)deltaCycles / 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_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... } - sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]); - - // currentPos is position from "zero" or soundBufferPos... - currentPos += soundBufferPos; - - while (soundBufferPos < currentPos) - soundBuffer[soundBufferPos++] = (uint8)sample; - - // This is done *after* in case the buffer had a long dead spot (I think...) - speakerState = !speakerState; - sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]); - lastToggleCycles = elapsedCycles; + soundBuffer[soundBufferPos++] = sample; +//WriteLog("WriteSampleToBuffer(): SDL_mutexV(mutex2)\n"); SDL_mutexV(mutex2); -// SDL_UnlockAudio(); } -void AddToSoundTimeBase(uint32 cycles) -{ - if (!soundInitialized) - return; - -// SDL_LockAudio(); - SDL_mutexP(mutex2); - sampleBase += (uint32)((double)cycles / CYCLES_PER_SAMPLE); - SDL_mutexV(mutex2); -// SDL_UnlockAudio(); -} -void AdjustLastToggleCycles(uint64 elapsedCycles) +void ToggleSpeaker(void) { -#if 0 - if (!soundInitialized) - return; - - SDL_mutexP(mutex2); - lastToggleCycles += elapsedCycles; - SDL_mutexV(mutex2); - -// We should also fill the buffer here as well, even if the speaker -// didn't toggle... !!! FIX !!! -#else -/* -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. -*/ -#warning "This is VERY similar to ToggleSpeaker(); merge into common function. !!! FIX !!!" if (!soundInitialized) return; -#ifdef DEBUG -printf("SOUND: AdjustLastToggleCycles() start...\n"); -#endif - // 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 - SDL_mutexP(mutex2); - while ((soundBufferPos + currentPos) > (SOUND_BUFFER_SIZE - 1)) - { - // Hm. - // This might not empty the buffer enough, causing hash and trash. !!! FIX !!! [DONE] - 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... - -//HMM, this doesn't need to lock or recalculate this value -// currentPos = (uint32)((double)deltaCycles / CYCLES_PER_SAMPLE); - } - - // Step 4: Backfill and adjust lastToggleCycles - // currentPos is position from "zero" or soundBufferPos... - currentPos += soundBufferPos; - - // Backfill with current toggle state - while (soundBufferPos < currentPos) - soundBuffer[soundBufferPos++] = (uint8)sample; - - SDL_mutexV(mutex2); - lastToggleCycles = elapsedCycles; -#ifdef DEBUG -printf("SOUND: AdjustLastToggleCycles() end...\n"); -#endif -#endif + speakerState = !speakerState; + sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]); } + 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()...) -*/ - - - - - - -