X-Git-Url: http://shamusworld.gotdns.org/cgi-bin/gitweb.cgi?a=blobdiff_plain;f=src%2Fsound.cpp;h=75f9ba2bae9779c3493752f28867905f59fd0d8d;hb=f36d026c7b8b398b88765ec5b67a3c767fe5fbad;hp=cd153df782fe1df8c89908318814fddc622e47fb;hpb=c0bc82a6324c65aefd62cc3eb33c09a14db638bf;p=apple2 diff --git a/src/sound.cpp b/src/sound.cpp old mode 100755 new mode 100644 index cd153df..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) @@ -61,6 +44,7 @@ // Local variables static SDL_AudioSpec desired, obtained; +static SDL_AudioDeviceID device; static bool soundInitialized = false; static bool speakerState = false; static int16_t soundBuffer[SOUND_BUFFER_SIZE]; @@ -70,7 +54,7 @@ static SDL_cond * conditional = NULL; static SDL_mutex * mutex = NULL; static SDL_mutex * mutex2 = NULL; static int16_t sample; -static uint8_t ampPtr = 14; // Start with -16 - +16 +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 @@ -81,6 +65,7 @@ static FILE * fp = NULL; static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length); + // // Initialize the SDL sound system // @@ -90,20 +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.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; @@ -116,7 +97,7 @@ 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"); @@ -125,6 +106,7 @@ return; #endif } + // // Close down the SDL sound subsystem // @@ -132,8 +114,8 @@ 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); @@ -145,11 +127,27 @@ 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 // -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.) @@ -159,29 +157,33 @@ 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: soundBufferPos = %i\n", soundBufferPos); SDL_mutexP(mutex2); // Recast this as a 16-bit type... 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) { + // The sound buffer is starved... for(uint32_t i=0; i (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_t sbpSave = soundBufferPos; -#endif - // Backfill with current toggle state - while (soundBufferPos < currentPos) - soundBuffer[soundBufferPos++] = (uint16_t)sample; - -#ifdef WRITE_OUT_WAVE - fwrite(&soundBuffer[sbpSave], sizeof(int16_t), currentPos - sbpSave, fp); -#endif - + soundBuffer[soundBufferPos++] = sample; +//WriteLog("WriteSampleToBuffer(): SDL_mutexV(mutex2)\n"); 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 8-bit samples - // Now 16 + // Currently set for 16-bit samples if (ampPtr < 16) ampPtr++; } + void VolumeDown(void) { if (ampPtr > 0) ampPtr--; } + 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()...) -*/ -