1) make sure AW does not fill the DS buffer, as semantics are unclear

2) process audio sooner, rather than a frame later
3) only 1 SDL buffer control (its size)
4) DS: Lock-Unlock probably reset the write cursor (in Windows)
5) DS: AW does not like DS buffer to be empty (or full??) at the begin

source/Speaker.cpp

@@ -569,8 +569,8 @@ static ULONG Spkr_SubmitWaveBuffer_FullSpeed(short* pSpeakerBuffer, ULONG nNumSa
 	UINT nBytesFree = g_dwDSSpkrBufferSize - nBytesRemaining;	// Calc free buffer space
 	ULONG nNumSamplesToUse = nNumSamples + nNumPadSamples;
 
-	if(nNumSamplesToUse * sizeof(short) > nBytesFree)
-		nNumSamplesToUse = nBytesFree / sizeof(short);
+	if(nNumSamplesToUse * sizeof(short) >= nBytesFree)
+		nNumSamplesToUse = nBytesFree / sizeof(short) - 1;
 
 	//
 
@@ -766,8 +766,8 @@ static ULONG Spkr_SubmitWaveBuffer(short* pSpeakerBuffer, ULONG nNumSamples)
 	UINT nBytesFree = g_dwDSSpkrBufferSize - nBytesRemaining;	// Calc free buffer space
 	ULONG nNumSamplesToUse = nNumSamples;
 
-	if(nNumSamplesToUse * sizeof(short) > nBytesFree)
-		nNumSamplesToUse = nBytesFree / sizeof(short);
+	if(nNumSamplesToUse * sizeof(short) >= nBytesFree)
+		nNumSamplesToUse = nBytesFree / sizeof(short) - 1;
 
 	if(bBufferError)
 		pSpeakerBuffer = &pSpeakerBuffer[nNumSamples - nNumSamplesToUse];

source/frontends/sdl/imgui/sdlsettings.cpp

@@ -500,14 +500,24 @@ namespace sa2
 
           ImGui::Separator();
 
-          if (ImGui::BeginTable("Devices", 5, ImGuiTableFlags_RowBg))
+          int size;
+          getAudioBufferSize(size);
+          if (ImGui::SliderInt("Buffer size", &size, 0, 1000, "%d ms"))
+          {
+            setAudioBufferSize(size);
+          }
+
+          ImGui::Separator();
+
+          if (ImGui::BeginTable("Devices", 6, ImGuiTableFlags_RowBg))
           {
             myAudioInfo = getAudioInfo();
             ImGui::TableSetupColumn("Running");
             ImGui::TableSetupColumn("Channels");
             ImGui::TableSetupColumn("Volume");
-            ImGui::TableSetupColumn("Buffer");
-            ImGui::TableSetupColumn("Queue");
+            ImGui::TableSetupColumn("Direct");
+            ImGui::TableSetupColumn("SDL");
+            ImGui::TableSetupColumn("Total");
             ImGui::TableHeadersRow();
 
             ImGui::PushItemFlag(ImGuiItemFlags_Disabled, true); // this requires imgui_internal.h
@@ -523,6 +533,9 @@ namespace sa2
               ImGui::SliderFloat("", &device.buffer, 0.0f, device.size, "%.3f");
               ImGui::TableNextColumn();
               ImGui::SliderFloat("", &device.queue, 0.0f, device.size, "%.3f");
+              ImGui::TableNextColumn();
+              float total = device.buffer + device.queue;
+              ImGui::SliderFloat("", &total, 0.0f, 2 * device.size, "%.3f");
             }
             ImGui::PushItemFlag(ImGuiItemFlags_Disabled, false);
 

source/frontends/sdl/main.cpp

@@ -140,13 +140,13 @@ void run_sdl(int argc, const char * argv [])
       frameTimer.tic();
 
       eventTimer.tic();
-      sa2::writeAudio();
       processEventsUthernet2(5);
       frame->ProcessEvents(quit);
       eventTimer.toc();
 
       cpuTimer.tic();
       frame->ExecuteOneFrame(oneFrame);
+      sa2::writeAudio();
       cpuTimer.toc();
 
       if (!options.headless)

source/frontends/sdl/sdirectsound.cpp

@@ -24,6 +24,8 @@ namespace
     void printInfo() const;
     sa2::SoundInfo getInfo() const;
 
+    static int ourBufferSize;
+
   private:
     IDirectSoundBuffer * myBuffer;
 
@@ -33,6 +35,7 @@ namespace
     SDL_AudioSpec myAudioSpec;
 
     int myBytesPerSecond;
+    int myBytesPerUnit;
 
     void close();
     bool isRunning() const;
@@ -41,6 +44,7 @@ namespace
     void mixBuffer(const void * ptr, const size_t size);
   };
 
+  int DirectSoundGenerator::ourBufferSize = 200;
 
   std::unordered_map<IDirectSoundBuffer *, std::shared_ptr<DirectSoundGenerator>> activeSoundGenerators;
 
@@ -48,6 +52,7 @@ namespace
     : myBuffer(buffer)
     , myAudioDevice(0)
     , myBytesPerSecond(0)
+    , myBytesPerUnit(0)
   {
     SDL_memset(&myAudioSpec, 0, sizeof(myAudioSpec));
   }
@@ -95,7 +100,8 @@ namespace
     if (myAudioDevice)
     {
       const int bitsPerSample = myAudioSpec.format & SDL_AUDIO_MASK_BITSIZE;
-      myBytesPerSecond = myAudioSpec.freq * myAudioSpec.channels * bitsPerSample / 8;
+      myBytesPerUnit = myAudioSpec.channels * bitsPerSample / 8;
+      myBytesPerSecond = myAudioSpec.freq * myBytesPerUnit;
 
       SDL_PauseAudioDevice(myAudioDevice, 0);
       return true;
@@ -133,7 +139,7 @@ namespace
       const float coeff = 1.0 / myBytesPerSecond;
       info.buffer = bytesInBuffer * coeff;
       info.queue = bytesInQueue * coeff;
-      info.size = myBuffer->bufferSize * coeff;
+      info.size = std::max(myBuffer->bufferSize * coeff, float(ourBufferSize / 1000.0));
     }
 
     return info;
@@ -177,14 +183,17 @@ namespace
     // otherwise AW starts generating a lot of samples
     // and we loose sync
 
-    // assume SDL's buffer is the same size as AW's
-    const int bytesFree = myBuffer->bufferSize - SDL_GetQueuedAudioSize(myAudioDevice);
+    const int queued = SDL_GetQueuedAudioSize(myAudioDevice);
+    const int bufferSize = myBytesPerSecond * DirectSoundGenerator::ourBufferSize / 1000;
 
-    if (bytesFree > 0)
+    if (queued < bufferSize)  // trigger is already <= target
     {
+      // make sure we only copy full frames
+      const int bytesToCopy = ((bufferSize - queued) / myBytesPerUnit) * myBytesPerUnit;
+
       LPVOID lpvAudioPtr1, lpvAudioPtr2;
       DWORD dwAudioBytes1, dwAudioBytes2;
-      myBuffer->Read(bytesFree, &lpvAudioPtr1, &dwAudioBytes1, &lpvAudioPtr2, &dwAudioBytes2);
+      myBuffer->Read(bytesToCopy, &lpvAudioPtr1, &dwAudioBytes1, &lpvAudioPtr2, &dwAudioBytes2);
 
       if (lpvAudioPtr1 && dwAudioBytes1)
       {
@@ -260,4 +269,14 @@ namespace sa2
     return info;
   }
 
+  void getAudioBufferSize(int & size)
+  {
+    size = DirectSoundGenerator::ourBufferSize;
+  }
+
+  void setAudioBufferSize(const int size)
+  {
+    DirectSoundGenerator::ourBufferSize = size;
+  }
+
 }

source/frontends/sdl/sdirectsound.h

@@ -22,4 +22,7 @@ namespace sa2
   void writeAudio();
   void printAudioInfo();
   std::vector<SoundInfo> getAudioInfo();
+
+  void getAudioBufferSize(int & size);
+  void setAudioBufferSize(const int size);
 }

source/linux/windows/dsound.cpp

@@ -55,11 +55,6 @@ HRESULT IDirectSoundBuffer::Stop()
   return DS_OK;
 }
 
-HRESULT IDirectSoundBuffer::SetCurrentPosition( DWORD dwNewPosition )
-{
-  return DS_OK;
-}
-
 HRESULT IDirectSoundBuffer::Play( DWORD dwReserved1, DWORD dwReserved2, DWORD dwFlags )
 {
   this->myStatus |= DSBSTATUS_PLAYING;
@@ -102,6 +97,7 @@ HRESULT IDirectSoundBuffer::Lock( DWORD dwWriteCursor, DWORD dwWriteBytes, LPVOI
   }
   else
   {
+    myWritePosition = dwWriteCursor;
     const DWORD availableInFirstPart = this->mySoundBuffer.size() - dwWriteCursor;
 
     *lplpvAudioPtr1 = this->mySoundBuffer.data() + dwWriteCursor;

source/linux/windows/dsound.h

@@ -78,8 +78,8 @@ class IDirectSoundBuffer : public IUnknown
   const std::unique_ptr<IDirectSoundNotify> mySoundNotify;
   std::vector<char> mySoundBuffer;
 
-  size_t myPlayPosition = 0;
-  size_t myWritePosition = 0;
+  size_t myPlayPosition  = 0x0000;
+  size_t myWritePosition = 0x0100;
   WORD myStatus = 0;
   LONG myVolume = DSBVOLUME_MAX;
 
@@ -95,7 +95,6 @@ class IDirectSoundBuffer : public IUnknown
 
   HRESULT QueryInterface(int riid, void **ppvObject);
 
-  HRESULT SetCurrentPosition( DWORD dwNewPosition );
   HRESULT GetCurrentPosition( LPDWORD lpdwCurrentPlayCursor, LPDWORD lpdwCurrentWriteCursor );
 
   // Read is NOT part of Windows API