Clarifying the OpenAL loop

because it isn't as clear as it can be

2 files changed, 99 insertions, 104 deletions

diff --git a/Source/Core/AudioCommon/Src/OpenALStream.cpp b/Source/Core/AudioCommon/Src/OpenALStream.cpp
index b1e601dfe7..754df6139a 100644
--- a/Source/Core/AudioCommon/Src/OpenALStream.cpp
+++ b/Source/Core/AudioCommon/Src/OpenALStream.cpp
@@ -97,7 +97,7 @@ void OpenALStream::SetVolume(int volume)
 	fVolume = (float)volume / 100.0f;
 
 	if (uiSource)
-		alSourcef(uiSource, AL_GAIN, fVolume); 
+		alSourcef(uiSource, AL_GAIN, fVolume);
 }
 
 void OpenALStream::Update()
@@ -124,6 +124,15 @@ void OpenALStream::SoundLoop()
 {
 	Common::SetCurrentThreadName("Audio thread - openal");
 
+	bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder;
+#if defined(__APPLE__)
+	bool float32_capable = false;
+	// OSX does not have the alext AL_FORMAT_51CHN32 yet.
+	surround_capable = false;
+#else
+	bool float32_capable = true;
+#endif
+
 	u32 ulFrequency = m_mixer->GetSampleRate();
 	numBuffers = Core::g_CoreStartupParameter.iLatency + 2; // OpenAL requires a minimum of two buffers
 
@@ -136,22 +145,20 @@ void OpenALStream::SoundLoop()
 	alGenSources(1, &uiSource);
 
 	// Short Silence
-	memset(sampleBuffer, 0, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * numBuffers);
-	memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * 4);
+	memset(sampleBuffer, 0, OAL_MAX_SAMPLES * numBuffers * FRAME_SURROUND_FLOAT);
+	memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * FRAME_STEREO_SHORT);
 	for (int i = 0; i < numBuffers; i++)
 	{
-#if !defined(__APPLE__)
-		if (Core::g_CoreStartupParameter.bDPL2Decoder)
-			alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency);
+		if (surround_capable)
+			alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * FRAME_SURROUND_FLOAT, ulFrequency);
 		else
-#endif
-			alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * 2 * 2, ulFrequency);
+			alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * FRAME_STEREO_SHORT, ulFrequency);
 	}
 	alSourceQueueBuffers(uiSource, numBuffers, uiBuffers);
 	alSourcePlay(uiSource);
-	
+
 	// Set the default sound volume as saved in the config file.
-	alSourcef(uiSource, AL_GAIN, fVolume); 
+	alSourcef(uiSource, AL_GAIN, fVolume);
 
 	// TODO: Error handling
 	//ALenum err = alGetError();
@@ -170,14 +177,7 @@ void OpenALStream::SoundLoop()
 	soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28);
 	soundTouch.setSetting(SETTING_OVERLAP_MS, 12);
 
-	bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder;
-#if defined(__APPLE__)
-	bool float32_capable = false;
-#else
-	bool float32_capable = true;
-#endif
-
-	while (!threadData) 
+	while (!threadData)
 	{
 		// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
 		const u32 stereo_16_bit_size = 4;
@@ -193,12 +193,9 @@ void OpenALStream::SoundLoop()
 		numSamples = m_mixer->Mix(realtimeBuffer, numSamples);
 
 		// Convert the samples from short to float
-		float dest[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS];
-		for (u32 i = 0; i < numSamples; ++i)
-		{
-			dest[i * 2 + 0] = (float)realtimeBuffer[i * 2 + 0] / (1 << 16);
-			dest[i * 2 + 1] = (float)realtimeBuffer[i * 2 + 1] / (1 << 16);
-		}
+		float dest[OAL_MAX_SAMPLES * STEREO_CHANNELS];
+		for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
+			dest[i] = (float)realtimeBuffer[i] / (1 << 16);
 
 		soundTouch.putSamples(dest, numSamples);
 
@@ -230,101 +227,94 @@ void OpenALStream::SoundLoop()
 				}
 			}
 
-			unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * SIZE_FLOAT * OAL_MAX_BUFFERS);
+			unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);
 
-			if (nSamples > minSamples)
+			if (nSamples < minSamples)
+				continue;
+
+			// Remove the Buffer from the Queue.  (uiBuffer contains the Buffer ID for the unqueued Buffer)
+			if (iBuffersFilled == 0)
 			{
-				// Remove the Buffer from the Queue.  (uiBuffer contains the Buffer ID for the unqueued Buffer)
-				if (iBuffersFilled == 0)
+				alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
+				ALenum err = alGetError();
+				if (err != 0)
 				{
-					alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
-					ALenum err = alGetError();
-					if (err != 0)
-					{
-						ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
-					}
+					ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
 				}
-#if defined(__APPLE__)
-				// OSX does not have the alext AL_FORMAT_51CHN32 yet.
-				surround_capable = false;
-#else
-				if (surround_capable)
+			}
+
+			if (surround_capable)
+			{
+				float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
+				dpl2decode(sampleBuffer, nSamples, dpl2);
+				alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
+				ALenum err = alGetError();
+				if (err == AL_INVALID_ENUM)
 				{
-					float dpl2[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
-					dpl2decode(sampleBuffer, nSamples, dpl2);
+					// 5.1 is not supported by the host, fallback to stereo
+					WARN_LOG(AUDIO, "Unable to set 5.1 surround mode.  Updating OpenAL Soft might fix this issue.");
+					surround_capable = false;
+				}
+				else if (err != 0)
+				{
+					ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err);
+				}
+			}
 
-					alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency);
+			else
+			{
+				if (float32_capable)
+				{
+					alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, nSamples * FRAME_STEREO_FLOAT, ulFrequency);
 					ALenum err = alGetError();
 					if (err == AL_INVALID_ENUM)
 					{
-						// 5.1 is not supported by the host, fallback to stereo
-						WARN_LOG(AUDIO, "Unable to set 5.1 surround mode.  Updating OpenAL Soft might fix this issue.");
-						surround_capable = false;
+						float32_capable = false;
 					}
 					else if (err != 0)
 					{
-						ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err);
-					}
-				}
-#endif
-				if (!surround_capable)
-				{
-#if !defined(__APPLE__)
-					if (float32_capable)
-					{
-						alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, nSamples * 4 * 2, ulFrequency);
-						ALenum err = alGetError();
-						if (err == AL_INVALID_ENUM)
-						{
-							float32_capable = false;
-						}
-						else if (err != 0)
-						{
-							ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
-						}
-					}
-#endif
-					if (!float32_capable)
-					{
-						// Convert the samples from float to short
-						short stereo[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS];
-						for (u32 i = 0; i < nSamples; ++i)
-						{
-							stereo[i * 2 + 0] = (short)((float)sampleBuffer[i * 2 + 0] * (1 << 16));
-							stereo[i * 2 + 1] = (short)((float)sampleBuffer[i * 2 + 1] * (1 << 16));
-						}
-						alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo, nSamples * 2 * 2, ulFrequency);
+						ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
 					}
 				}
 
-				alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
-				ALenum err = alGetError();
-				if (err != 0)
+				else
 				{
-					ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
+					// Convert the samples from float to short
+					short stereo[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
+					for (u32 i = 0; i < nSamples * STEREO_CHANNELS; ++i)
+						stereo[i] = (short)((float)sampleBuffer[i] * (1 << 16));
+
+					alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo, nSamples * FRAME_STEREO_SHORT, ulFrequency);
 				}
-				iBuffersFilled++;
+			}
+
+			alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
+			ALenum err = alGetError();
+			if (err != 0)
+			{
+				ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
+			}
+			iBuffersFilled++;
 
-				if (iBuffersFilled == numBuffers)
+			if (iBuffersFilled == numBuffers)
+			{
+				alSourcePlay(uiSource);
+				err = alGetError();
+				if (err != 0)
 				{
-					alSourcePlay(uiSource);
-					err = alGetError();
-					if (err != 0)
-					{
-						ERROR_LOG(AUDIO, "Error occurred during playback: %08x", err);
-					}
+					ERROR_LOG(AUDIO, "Error occurred during playback: %08x", err);
 				}
+			}
 
-				alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
-				if (iState != AL_PLAYING)
+			alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
+			if (iState != AL_PLAYING)
+			{
+				// Buffer underrun occurred, resume playback
+				alSourcePlay(uiSource);
+				err = alGetError();
+				if (err != 0)
 				{
-					// Buffer underrun occurred, resume playback
-					alSourcePlay(uiSource);
-					err = alGetError();
-					if (err != 0)
-					{
-						ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
-					}
+					ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
 				}
 			}
 		}
diff --git a/Source/Core/AudioCommon/Src/OpenALStream.h b/Source/Core/AudioCommon/Src/OpenALStream.h
index 42371db045..24217169bd 100644
--- a/Source/Core/AudioCommon/Src/OpenALStream.h
+++ b/Source/Core/AudioCommon/Src/OpenALStream.h
@@ -30,11 +30,16 @@
 #include <soundtouch/STTypes.h>
 
 // 16 bit Stereo
-#define SFX_MAX_SOURCE		1
-#define OAL_MAX_BUFFERS		32
-#define OAL_MAX_SAMPLES		256
-#define SURROUND_CHANNELS	6	// number of channels in surround mode
-#define SIZE_FLOAT			4   // size of a float in bytes
+#define SFX_MAX_SOURCE			1
+#define OAL_MAX_BUFFERS			32
+#define OAL_MAX_SAMPLES			256
+#define STEREO_CHANNELS			2
+#define SURROUND_CHANNELS		6	// number of channels in surround mode
+#define SIZE_SHORT				2
+#define SIZE_FLOAT				4   // size of a float in bytes
+#define FRAME_STEREO_SHORT		STEREO_CHANNELS * SIZE_SHORT
+#define FRAME_STEREO_FLOAT		STEREO_CHANNELS * SIZE_FLOAT
+#define FRAME_SURROUND_FLOAT	SURROUND_CHANNELS * SIZE_FLOAT
 #endif
 
 class OpenALStream: public SoundStream
@@ -61,8 +66,8 @@ private:
 	std::thread thread;
 	Common::Event soundSyncEvent;
 
-	short realtimeBuffer[OAL_MAX_SAMPLES * 2];
-	soundtouch::SAMPLETYPE sampleBuffer[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
+	short realtimeBuffer[OAL_MAX_SAMPLES * STEREO_CHANNELS];
+	soundtouch::SAMPLETYPE sampleBuffer[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
 	ALuint uiBuffers[OAL_MAX_BUFFERS];
 	ALuint uiSource;
 	ALfloat fVolume;