Conversion of V65C02 to pointers, misc. whitespace cleanups.

[apple2] / src / sound.cpp

diff --git a/src/sound.cpp b/src/sound.cpp
index df2de2b4ab59d3c64b80cbc49eb8002ee68734cf..67b3090b6be2efa9c17f1f64a2efc4a7678fd74d 100644 (file)
--- a/src/sound.cpp
+++ b/src/sound.cpp
@@ -2,7 +2,7 @@
 // Sound Interface
 //
 // by James Hammons
-// (C) 2005 Underground Software
+// (C) 2005-2018 Underground Software
 //
 // JLH = James Hammons <jlhamm@acm.org>
 //
@@ -47,10 +47,6 @@ static bool soundInitialized = false;
 static bool speakerState = false;
 static uint16_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 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,
@@ -61,6 +57,10 @@ static int16_t amplitude[17] = { 0, 1, 2, 3, 7, 15, 31, 63, 127, 255,
 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
 //
@@ -81,14 +81,10 @@ void SoundInit(void)
                return;
        }
 
-//     conditional = SDL_CreateCond();
-//     mutex = SDL_CreateMutex();
-//     mutex2 = SDL_CreateMutex();// Let's try real signalling...
        soundBufferPos = 0;
-//     lastToggleCycles = 0;
-       sample = desired.silence;       // ? wilwok ? yes
+       sample = desired.silence;               // ? wilwok ? yes
 
-       SDL_PauseAudioDevice(device, 0);                        // Start playback!
+       SDL_PauseAudioDevice(device, 0);// Start playback!
        soundInitialized = true;
        WriteLog("Sound: Successfully initialized.\n");
 }
@@ -103,9 +99,6 @@ void SoundDone(void)
        {
                SDL_PauseAudioDevice(device, 1);
                SDL_CloseAudioDevice(device);
-//             SDL_DestroyCond(conditional);
-//             SDL_DestroyMutex(mutex);
-//             SDL_DestroyMutex(mutex2);
                WriteLog("Sound: Done.\n");
        }
 }
@@ -133,18 +126,11 @@ static uint32_t lastStarve = 0;
 static void SDLSoundCallback(void * /*userdata*/, Uint8 * buffer8, int length8)
 {
 sndFrmCnt++;
-//WriteLog("SDLSoundCallback(): begin (soundBufferPos=%i)\n", soundBufferPos);
-
-       // 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...
        uint16_t * buffer = (uint16_t *)buffer8;
        uint32_t length = (uint32_t)length8 / 2;
 
-//WriteLog("SDLSoundCallback(): filling buffer...\n");
        if (soundBufferPos < length)
        {
 WriteLog("*** Sound buffer starved (%d short) *** [%d delta %d]\n", length - soundBufferPos, sndFrmCnt, sndFrmCnt - lastStarve);
@@ -177,17 +163,12 @@ lastStarve = sndFrmCnt;
                for(uint32_t i=0; i<soundBufferPos; i++)
                        soundBuffer[i] = soundBuffer[length + i];
        }
-
-       // Free the mutex...
-//WriteLog("SDLSoundCallback(): SDL_mutexV(mutex2)\n");
-//     SDL_mutexV(mutex2);
-       // Wake up any threads waiting for the buffer to drain...
-//     SDL_CondSignal(conditional);
-//WriteLog("SDLSoundCallback(): end\n");
 }
 
 
+//
 // This is called by the main CPU thread every ~21.666 cycles.
+//
 void WriteSampleToBuffer(void)
 {
 #ifdef USE_NEW_AY8910
@@ -202,31 +183,17 @@ void WriteSampleToBuffer(void)
        int16_t adjustedMockingboard = (s1 / 8) + (s2 / 8) + (s3 / 8)
                + (s4 / 8) + (s5 / 8) + (s6 / 8);
 #endif
-//need to do this *before* mixing, as by this time, it's too late and the sample is probably already oversaturated
-//     adjustedMockingboard /= 8;
-
-//WriteLog("WriteSampleToBuffer(): SDL_mutexP(mutex2)\n");
-//     SDL_mutexP(mutex2);
 
        // This should almost never happen, but, if it does...
        while (soundBufferPos >= (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...
                SDL_Delay(1);
        }
 
        SDL_LockAudioDevice(device);
        soundBuffer[soundBufferPos++] = sample + adjustedMockingboard;
        SDL_UnlockAudioDevice(device);
-
-//     soundBuffer[soundBufferPos++] = sample;
-//WriteLog("WriteSampleToBuffer(): SDL_mutexV(mutex2)\n");
-//     SDL_mutexV(mutex2);
 }
 
 
@@ -236,7 +203,7 @@ void ToggleSpeaker(void)
                return;
 
        speakerState = !speakerState;
-       sample = (speakerState ? amplitude[ampPtr] : 0);//-amplitude[ampPtr]);
+       sample = (speakerState ? amplitude[ampPtr] : 0);
 }