X-Git-Url: http://shamusworld.gotdns.org/cgi-bin/gitweb.cgi?a=blobdiff_plain;f=src%2Fsound.cpp;h=4cab23ec0aad0ec7a353e97869cf6ed53d1e5179;hb=c0001155bc0909da61f6c849c0be9b16e9b7f4b6;hp=cd153df782fe1df8c89908318814fddc622e47fb;hpb=c0bc82a6324c65aefd62cc3eb33c09a14db638bf;p=apple2 diff --git a/src/sound.cpp b/src/sound.cpp index cd153df..4cab23e 100755 --- a/src/sound.cpp +++ b/src/sound.cpp @@ -61,6 +61,7 @@ // 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 +71,7 @@ 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 +82,7 @@ static FILE * fp = NULL; static void SDLSoundCallback(void * userdata, Uint8 * buffer, int length); + // // Initialize the SDL sound system // @@ -90,20 +92,16 @@ 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 +114,7 @@ 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 +123,7 @@ return; #endif } + // // Close down the SDL sound subsystem // @@ -132,8 +131,10 @@ void SoundDone(void) { if (soundInitialized) { - SDL_PauseAudio(true); - SDL_CloseAudio(); +// SDL_PauseAudio(true); + SDL_PauseAudioDevice(device, 1); +// SDL_CloseAudio(); + SDL_CloseAudioDevice(device); SDL_DestroyCond(conditional); SDL_DestroyMutex(mutex); SDL_DestroyMutex(mutex2); @@ -145,11 +146,27 @@ 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.) @@ -159,12 +176,14 @@ 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; +//WriteLog("SDLSoundCallback(): filling buffer...\n"); 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); + 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... + } + + soundBuffer[soundBufferPos++] = sample; +//WriteLog("WriteSampleToBuffer(): SDL_mutexV(mutex2)\n"); + SDL_mutexV(mutex2); +} + + // Need some interface functions here to take care of flipping the // waveform at the correct time in the sound stream... @@ -222,6 +269,7 @@ void HandleBuffer(uint64_t elapsedCycles) // 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, @@ -240,7 +288,7 @@ void HandleBuffer(uint64_t elapsedCycles) #endif // Backfill with current toggle state while (soundBufferPos < currentPos) - soundBuffer[soundBufferPos++] = (uint16_t)sample; + soundBuffer[soundBufferPos++] = sample; #ifdef WRITE_OUT_WAVE fwrite(&soundBuffer[sbpSave], sizeof(int16_t), currentPos - sbpSave, fp); @@ -250,16 +298,18 @@ void HandleBuffer(uint64_t elapsedCycles) lastToggleCycles = elapsedCycles; } + void ToggleSpeaker(uint64_t elapsedCycles) { if (!soundInitialized) return; - HandleBuffer(elapsedCycles); +// HandleBuffer(elapsedCycles); speakerState = !speakerState; sample = (speakerState ? amplitude[ampPtr] : -amplitude[ampPtr]); } + void AdjustLastToggleCycles(uint64_t elapsedCycles) { if (!soundInitialized) @@ -292,20 +342,22 @@ 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;