X-Git-Url: http://shamusworld.gotdns.org/cgi-bin/gitweb.cgi?a=blobdiff_plain;f=src%2Fsound.cpp;h=75f9ba2bae9779c3493752f28867905f59fd0d8d;hb=f36d026c7b8b398b88765ec5b67a3c767fe5fbad;hp=fd6754d7bc0c8c7e4483e652319df22efc5c3abf;hpb=a2e007c1e012426f3fe8bc48bf8d6ee934420214;p=apple2 diff --git a/src/sound.cpp b/src/sound.cpp old mode 100755 new mode 100644 index fd6754d..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 // --- ---------- ------------------------------------------------------------ @@ -17,7 +17,7 @@ // 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" @@ -31,27 +31,10 @@ //#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) @@ -131,9 +114,7 @@ void SoundDone(void) { if (soundInitialized) { -// SDL_PauseAudio(true); SDL_PauseAudioDevice(device, 1); -// SDL_CloseAudio(); SDL_CloseAudioDevice(device); SDL_DestroyCond(conditional); SDL_DestroyMutex(mutex); @@ -147,6 +128,20 @@ void SoundDone(void) } +void SoundPause(void) +{ + if (soundInitialized) + SDL_PauseAudioDevice(device, 1); +} + + +void SoundResume(void) +{ + if (soundInitialized) + SDL_PauseAudioDevice(device, 0); +} + + // // Sound card callback handler // @@ -162,7 +157,7 @@ static void SDLSoundCallback(void * /*userdata*/, Uint8 * buffer8, int length8) // Let's try using a mutex for shared resource consumption... //Actually, I think Lock/UnlockAudio() does this already... -//WriteLog("SDLSoundCallback(): SDL_mutexP(mutex2)\n"); +//WriteLog("SDLSoundCallback: soundBufferPos = %i\n", soundBufferPos); SDL_mutexP(mutex2); // Recast this as a 16-bit type... @@ -170,22 +165,22 @@ static void SDLSoundCallback(void * /*userdata*/, Uint8 * buffer8, int length8) uint32_t length = (uint32_t)length8 / 2; //WriteLog("SDLSoundCallback(): filling buffer...\n"); - if (soundBufferPos < length) // The sound buffer is starved... + if (soundBufferPos < length) { + // The sound buffer is starved... for(uint32_t i=0; i= (SOUND_BUFFER_SIZE - 1)) { //WriteLog("WriteSampleToBuffer(): Waiting for sound thread. soundBufferPos=%i, SOUNDBUFFERSIZE-1=%i\n", soundBufferPos, SOUND_BUFFER_SIZE-1); @@ -228,107 +223,16 @@ void WriteSampleToBuffer(void) } -// 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_t elapsedCycles) -{ - // Step 1: Calculate delta time - uint64_t deltaCycles = elapsedCycles - lastToggleCycles; - - // Step 2: Calculate new buffer position - uint32_t currentPos = (uint32_t)((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)) - { - 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_t sbpSave = soundBufferPos; -#endif - // Backfill with current toggle state - while (soundBufferPos < currentPos) - soundBuffer[soundBufferPos++] = sample; - -#ifdef WRITE_OUT_WAVE - fwrite(&soundBuffer[sbpSave], sizeof(int16_t), currentPos - sbpSave, fp); -#endif - - SDL_mutexV(mutex2); - lastToggleCycles = elapsedCycles; -} - - -void ToggleSpeaker(uint64_t elapsedCycles) +void ToggleSpeaker(void) { if (!soundInitialized) return; -// HandleBuffer(elapsedCycles); speakerState = !speakerState; sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]); } -void AdjustLastToggleCycles(uint64_t 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 16-bit samples @@ -349,66 +253,3 @@ 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()...) -*/ -