Reformat all the things. Have fun with merge conflicts.

1 files changed, 328 insertions, 316 deletions

diff --git a/Source/Core/AudioCommon/OpenALStream.cpp b/Source/Core/AudioCommon/OpenALStream.cpp
index b99e7e5bf7..9f0ea2af57 100644
--- a/Source/Core/AudioCommon/OpenALStream.cpp
+++ b/Source/Core/AudioCommon/OpenALStream.cpp
@@ -5,11 +5,11 @@
 #include <cstring>
 #include <thread>
 
-#include "AudioCommon/aldlist.h"
 #include "AudioCommon/DPL2Decoder.h"
 #include "AudioCommon/OpenALStream.h"
-#include "Common/Thread.h"
+#include "AudioCommon/aldlist.h"
 #include "Common/Logging/Log.h"
+#include "Common/Thread.h"
 #include "Core/ConfigManager.h"
 
 #if defined HAVE_OPENAL && HAVE_OPENAL
@@ -25,351 +25,363 @@ static soundtouch::SoundTouch soundTouch;
 //
 bool OpenALStream::Start()
 {
-	m_run_thread.store(true);
-	bool bReturn = false;
-
-	ALDeviceList pDeviceList;
-	if (pDeviceList.GetNumDevices())
-	{
-		char *defDevName = pDeviceList.GetDeviceName(pDeviceList.GetDefaultDevice());
-
-		WARN_LOG(AUDIO, "Found OpenAL device %s", defDevName);
-
-		ALCdevice *pDevice = alcOpenDevice(defDevName);
-		if (pDevice)
-		{
-			ALCcontext *pContext = alcCreateContext(pDevice, nullptr);
-			if (pContext)
-			{
-				// Used to determine an appropriate period size (2x period = total buffer size)
-				//ALCint refresh;
-				//alcGetIntegerv(pDevice, ALC_REFRESH, 1, &refresh);
-				//period_size_in_millisec = 1000 / refresh;
-
-				alcMakeContextCurrent(pContext);
-				thread = std::thread(&OpenALStream::SoundLoop, this);
-				bReturn = true;
-			}
-			else
-			{
-				alcCloseDevice(pDevice);
-				PanicAlertT("OpenAL: can't create context for device %s", defDevName);
-			}
-		}
-		else
-		{
-			PanicAlertT("OpenAL: can't open device %s", defDevName);
-		}
-	}
-	else
-	{
-		PanicAlertT("OpenAL: can't find sound devices");
-	}
-
-	// Initialize DPL2 parameters
-	DPL2Reset();
-
-	soundTouch.clear();
-	return bReturn;
+  m_run_thread.store(true);
+  bool bReturn = false;
+
+  ALDeviceList pDeviceList;
+  if (pDeviceList.GetNumDevices())
+  {
+    char* defDevName = pDeviceList.GetDeviceName(pDeviceList.GetDefaultDevice());
+
+    WARN_LOG(AUDIO, "Found OpenAL device %s", defDevName);
+
+    ALCdevice* pDevice = alcOpenDevice(defDevName);
+    if (pDevice)
+    {
+      ALCcontext* pContext = alcCreateContext(pDevice, nullptr);
+      if (pContext)
+      {
+        // Used to determine an appropriate period size (2x period = total buffer size)
+        // ALCint refresh;
+        // alcGetIntegerv(pDevice, ALC_REFRESH, 1, &refresh);
+        // period_size_in_millisec = 1000 / refresh;
+
+        alcMakeContextCurrent(pContext);
+        thread = std::thread(&OpenALStream::SoundLoop, this);
+        bReturn = true;
+      }
+      else
+      {
+        alcCloseDevice(pDevice);
+        PanicAlertT("OpenAL: can't create context for device %s", defDevName);
+      }
+    }
+    else
+    {
+      PanicAlertT("OpenAL: can't open device %s", defDevName);
+    }
+  }
+  else
+  {
+    PanicAlertT("OpenAL: can't find sound devices");
+  }
+
+  // Initialize DPL2 parameters
+  DPL2Reset();
+
+  soundTouch.clear();
+  return bReturn;
 }
 
 void OpenALStream::Stop()
 {
-	m_run_thread.store(false);
-	// kick the thread if it's waiting
-	soundSyncEvent.Set();
+  m_run_thread.store(false);
+  // kick the thread if it's waiting
+  soundSyncEvent.Set();
 
-	soundTouch.clear();
+  soundTouch.clear();
 
-	thread.join();
+  thread.join();
 
-	alSourceStop(uiSource);
-	alSourcei(uiSource, AL_BUFFER, 0);
+  alSourceStop(uiSource);
+  alSourcei(uiSource, AL_BUFFER, 0);
 
-	// Clean up buffers and sources
-	alDeleteSources(1, &uiSource);
-	uiSource = 0;
-	alDeleteBuffers(numBuffers, uiBuffers);
+  // Clean up buffers and sources
+  alDeleteSources(1, &uiSource);
+  uiSource = 0;
+  alDeleteBuffers(numBuffers, uiBuffers);
 
-	ALCcontext *pContext = alcGetCurrentContext();
-	ALCdevice *pDevice = alcGetContextsDevice(pContext);
+  ALCcontext* pContext = alcGetCurrentContext();
+  ALCdevice* pDevice = alcGetContextsDevice(pContext);
 
-	alcMakeContextCurrent(nullptr);
-	alcDestroyContext(pContext);
-	alcCloseDevice(pDevice);
+  alcMakeContextCurrent(nullptr);
+  alcDestroyContext(pContext);
+  alcCloseDevice(pDevice);
 }
 
 void OpenALStream::SetVolume(int volume)
 {
-	fVolume = (float)volume / 100.0f;
+  fVolume = (float)volume / 100.0f;
 
-	if (uiSource)
-		alSourcef(uiSource, AL_GAIN, fVolume);
+  if (uiSource)
+    alSourcef(uiSource, AL_GAIN, fVolume);
 }
 
 void OpenALStream::Update()
 {
-	soundSyncEvent.Set();
+  soundSyncEvent.Set();
 }
 
 void OpenALStream::Clear(bool mute)
 {
-	m_muted = mute;
-
-	if (m_muted)
-	{
-		soundTouch.clear();
-		alSourceStop(uiSource);
-	}
-	else
-	{
-		alSourcePlay(uiSource);
-	}
+  m_muted = mute;
+
+  if (m_muted)
+  {
+    soundTouch.clear();
+    alSourceStop(uiSource);
+  }
+  else
+  {
+    alSourcePlay(uiSource);
+  }
 }
 
 void OpenALStream::SoundLoop()
 {
-	Common::SetCurrentThreadName("Audio thread - openal");
+  Common::SetCurrentThreadName("Audio thread - openal");
 
-	bool surround_capable = SConfig::GetInstance().bDPL2Decoder;
+  bool surround_capable = SConfig::GetInstance().bDPL2Decoder;
 #if defined(__APPLE__)
-	bool float32_capable = false;
-	const ALenum AL_FORMAT_STEREO_FLOAT32 = 0;
-	// OS X does not have the alext AL_FORMAT_51CHN32 yet.
-	surround_capable = false;
-	const ALenum AL_FORMAT_51CHN32 = 0;
-	const ALenum AL_FORMAT_51CHN16 = 0;
+  bool float32_capable = false;
+  const ALenum AL_FORMAT_STEREO_FLOAT32 = 0;
+  // OS X does not have the alext AL_FORMAT_51CHN32 yet.
+  surround_capable = false;
+  const ALenum AL_FORMAT_51CHN32 = 0;
+  const ALenum AL_FORMAT_51CHN16 = 0;
 #else
-	bool float32_capable = true;
+  bool float32_capable = true;
 #endif
 
-	u32 ulFrequency = m_mixer->GetSampleRate();
-	numBuffers = SConfig::GetInstance().iLatency + 2; // OpenAL requires a minimum of two buffers
-
-	memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
-	uiSource = 0;
-
-	// Checks if a X-Fi is being used. If it is, disable FLOAT32 support as this sound card has no support for it even though it reports it does.
-	if (strstr(alGetString(AL_RENDERER), "X-Fi"))
-		float32_capable = false;
-
-	// Generate some AL Buffers for streaming
-	alGenBuffers(numBuffers, (ALuint *)uiBuffers);
-	// Generate a Source to playback the Buffers
-	alGenSources(1, &uiSource);
-
-	// Short Silence
-	if (float32_capable)
-		memset(sampleBuffer, 0, OAL_MAX_SAMPLES * numBuffers * FRAME_SURROUND_FLOAT);
-	else
-		memset(sampleBuffer, 0, OAL_MAX_SAMPLES * numBuffers * FRAME_SURROUND_SHORT);
-
-	memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * FRAME_STEREO_SHORT);
-
-	for (int i = 0; i < numBuffers; i++)
-	{
-		if (surround_capable)
-		{
-			if (float32_capable)
-				alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * FRAME_SURROUND_FLOAT, ulFrequency);
-			else
-				alBufferData(uiBuffers[i], AL_FORMAT_51CHN16, sampleBuffer, 4 * FRAME_SURROUND_SHORT, ulFrequency);
-		}
-		else
-		{
-			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);
-
-	// TODO: Error handling
-	//ALenum err = alGetError();
-
-	ALint iBuffersFilled = 0;
-	ALint iBuffersProcessed = 0;
-	ALint iState = 0;
-	ALuint uiBufferTemp[OAL_MAX_BUFFERS] = { 0 };
-
-	soundTouch.setChannels(2);
-	soundTouch.setSampleRate(ulFrequency);
-	soundTouch.setTempo(1.0);
-	soundTouch.setSetting(SETTING_USE_QUICKSEEK, 0);
-	soundTouch.setSetting(SETTING_USE_AA_FILTER, 0);
-	soundTouch.setSetting(SETTING_SEQUENCE_MS, 1);
-	soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28);
-	soundTouch.setSetting(SETTING_OVERLAP_MS, 12);
-
-	while (m_run_thread.load())
-	{
-		// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
-		const u32 stereo_16_bit_size = 4;
-		const u32 dma_length = 32;
-		const u64 ais_samples_per_second = 48000 * stereo_16_bit_size;
-		u64 audio_dma_period = SystemTimers::GetTicksPerSecond() / (AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length);
-		u64 num_samples_to_render = (audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond();
-
-		unsigned int numSamples = (unsigned int)num_samples_to_render;
-		unsigned int minSamples = surround_capable ? 240 : 0; // DPL2 accepts 240 samples minimum (FWRDURATION)
-
-		numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
-		numSamples = m_mixer->Mix(realtimeBuffer, numSamples, false);
-
-		// Convert the samples from short to float
-		float dest[OAL_MAX_SAMPLES * STEREO_CHANNELS];
-		for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
-			dest[i] = (float)realtimeBuffer[i] / (1 << 15);
-
-		soundTouch.putSamples(dest, numSamples);
-
-		if (iBuffersProcessed == iBuffersFilled)
-		{
-			alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed);
-			iBuffersFilled = 0;
-		}
-
-		if (iBuffersProcessed)
-		{
-			double rate = (double)m_mixer->GetCurrentSpeed();
-			if (rate <= 0)
-			{
-				Core::RequestRefreshInfo();
-				rate = (double)m_mixer->GetCurrentSpeed();
-			}
-
-			// Place a lower limit of 10% speed.  When a game boots up, there will be
-			// many silence samples.  These do not need to be timestretched.
-			if (rate > 0.10)
-			{
-				soundTouch.setTempo(rate);
-				if (rate > 10)
-				{
-					soundTouch.clear();
-				}
-			}
-
-			unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);
-
-			if (nSamples <= minSamples)
-				continue;
-
-			// 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)
-				{
-					ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
-				}
-			}
-
-			if (surround_capable)
-			{
-				float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
-				DPL2Decode(sampleBuffer, nSamples, dpl2);
-
-				// zero-out the subwoofer channel - DPL2Decode generates a pretty
-				// good 5.0 but not a good 5.1 output.  Sadly there is not a 5.0
-				// AL_FORMAT_50CHN32 to make this super-explicit.
-				// DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
-				for (u32 i = 0; i < nSamples; ++i)
-				{
-					dpl2[i*SURROUND_CHANNELS + 3 /*sub/lfe*/] = 0.0f;
-				}
-
-				if (float32_capable)
-				{
-					alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
-				}
-				else
-				{
-					short surround_short[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
-					for (u32 i = 0; i < nSamples * SURROUND_CHANNELS; ++i)
-						surround_short[i] = (short)((float)dpl2[i] * (1 << 15));
-
-					alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN16, surround_short, nSamples * FRAME_SURROUND_SHORT, 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;
-				}
-				else if (err != 0)
-				{
-					ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err);
-				}
-			}
-
-			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)
-					{
-						float32_capable = false;
-					}
-					else if (err != 0)
-					{
-						ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
-					}
-				}
-
-				else
-				{
-					// 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 << 15));
-
-					alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo, nSamples * FRAME_STEREO_SHORT, ulFrequency);
-				}
-			}
-
-			alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
-			ALenum err = alGetError();
-			if (err != 0)
-			{
-				ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
-			}
-			iBuffersFilled++;
-
-			if (iBuffersFilled == numBuffers)
-			{
-				alSourcePlay(uiSource);
-				err = alGetError();
-				if (err != 0)
-				{
-					ERROR_LOG(AUDIO, "Error occurred during playback: %08x", err);
-				}
-			}
-
-			alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
-			if (iState != AL_PLAYING)
-			{
-				// Buffer underrun occurred, resume playback
-				alSourcePlay(uiSource);
-				err = alGetError();
-				if (err != 0)
-				{
-					ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
-				}
-			}
-		}
-		else
-		{
-			soundSyncEvent.Wait();
-		}
-	}
+  u32 ulFrequency = m_mixer->GetSampleRate();
+  numBuffers = SConfig::GetInstance().iLatency + 2;  // OpenAL requires a minimum of two buffers
+
+  memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
+  uiSource = 0;
+
+  // Checks if a X-Fi is being used. If it is, disable FLOAT32 support as this sound card has no
+  // support for it even though it reports it does.
+  if (strstr(alGetString(AL_RENDERER), "X-Fi"))
+    float32_capable = false;
+
+  // Generate some AL Buffers for streaming
+  alGenBuffers(numBuffers, (ALuint*)uiBuffers);
+  // Generate a Source to playback the Buffers
+  alGenSources(1, &uiSource);
+
+  // Short Silence
+  if (float32_capable)
+    memset(sampleBuffer, 0, OAL_MAX_SAMPLES * numBuffers * FRAME_SURROUND_FLOAT);
+  else
+    memset(sampleBuffer, 0, OAL_MAX_SAMPLES * numBuffers * FRAME_SURROUND_SHORT);
+
+  memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * FRAME_STEREO_SHORT);
+
+  for (int i = 0; i < numBuffers; i++)
+  {
+    if (surround_capable)
+    {
+      if (float32_capable)
+        alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * FRAME_SURROUND_FLOAT,
+                     ulFrequency);
+      else
+        alBufferData(uiBuffers[i], AL_FORMAT_51CHN16, sampleBuffer, 4 * FRAME_SURROUND_SHORT,
+                     ulFrequency);
+    }
+    else
+    {
+      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);
+
+  // TODO: Error handling
+  // ALenum err = alGetError();
+
+  ALint iBuffersFilled = 0;
+  ALint iBuffersProcessed = 0;
+  ALint iState = 0;
+  ALuint uiBufferTemp[OAL_MAX_BUFFERS] = {0};
+
+  soundTouch.setChannels(2);
+  soundTouch.setSampleRate(ulFrequency);
+  soundTouch.setTempo(1.0);
+  soundTouch.setSetting(SETTING_USE_QUICKSEEK, 0);
+  soundTouch.setSetting(SETTING_USE_AA_FILTER, 0);
+  soundTouch.setSetting(SETTING_SEQUENCE_MS, 1);
+  soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28);
+  soundTouch.setSetting(SETTING_OVERLAP_MS, 12);
+
+  while (m_run_thread.load())
+  {
+    // num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
+    const u32 stereo_16_bit_size = 4;
+    const u32 dma_length = 32;
+    const u64 ais_samples_per_second = 48000 * stereo_16_bit_size;
+    u64 audio_dma_period = SystemTimers::GetTicksPerSecond() /
+                           (AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length);
+    u64 num_samples_to_render =
+        (audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond();
+
+    unsigned int numSamples = (unsigned int)num_samples_to_render;
+    unsigned int minSamples =
+        surround_capable ? 240 : 0;  // DPL2 accepts 240 samples minimum (FWRDURATION)
+
+    numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
+    numSamples = m_mixer->Mix(realtimeBuffer, numSamples, false);
+
+    // Convert the samples from short to float
+    float dest[OAL_MAX_SAMPLES * STEREO_CHANNELS];
+    for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
+      dest[i] = (float)realtimeBuffer[i] / (1 << 15);
+
+    soundTouch.putSamples(dest, numSamples);
+
+    if (iBuffersProcessed == iBuffersFilled)
+    {
+      alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed);
+      iBuffersFilled = 0;
+    }
+
+    if (iBuffersProcessed)
+    {
+      double rate = (double)m_mixer->GetCurrentSpeed();
+      if (rate <= 0)
+      {
+        Core::RequestRefreshInfo();
+        rate = (double)m_mixer->GetCurrentSpeed();
+      }
+
+      // Place a lower limit of 10% speed.  When a game boots up, there will be
+      // many silence samples.  These do not need to be timestretched.
+      if (rate > 0.10)
+      {
+        soundTouch.setTempo(rate);
+        if (rate > 10)
+        {
+          soundTouch.clear();
+        }
+      }
+
+      unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);
+
+      if (nSamples <= minSamples)
+        continue;
+
+      // 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)
+        {
+          ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
+        }
+      }
+
+      if (surround_capable)
+      {
+        float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
+        DPL2Decode(sampleBuffer, nSamples, dpl2);
+
+        // zero-out the subwoofer channel - DPL2Decode generates a pretty
+        // good 5.0 but not a good 5.1 output.  Sadly there is not a 5.0
+        // AL_FORMAT_50CHN32 to make this super-explicit.
+        // DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
+        for (u32 i = 0; i < nSamples; ++i)
+        {
+          dpl2[i * SURROUND_CHANNELS + 3 /*sub/lfe*/] = 0.0f;
+        }
+
+        if (float32_capable)
+        {
+          alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2,
+                       nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
+        }
+        else
+        {
+          short surround_short[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
+          for (u32 i = 0; i < nSamples * SURROUND_CHANNELS; ++i)
+            surround_short[i] = (short)((float)dpl2[i] * (1 << 15));
+
+          alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN16, surround_short,
+                       nSamples * FRAME_SURROUND_SHORT, 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;
+        }
+        else if (err != 0)
+        {
+          ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err);
+        }
+      }
+
+      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)
+          {
+            float32_capable = false;
+          }
+          else if (err != 0)
+          {
+            ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
+          }
+        }
+
+        else
+        {
+          // 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 << 15));
+
+          alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo,
+                       nSamples * FRAME_STEREO_SHORT, ulFrequency);
+        }
+      }
+
+      alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
+      ALenum err = alGetError();
+      if (err != 0)
+      {
+        ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
+      }
+      iBuffersFilled++;
+
+      if (iBuffersFilled == numBuffers)
+      {
+        alSourcePlay(uiSource);
+        err = alGetError();
+        if (err != 0)
+        {
+          ERROR_LOG(AUDIO, "Error occurred during playback: %08x", err);
+        }
+      }
+
+      alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
+      if (iState != AL_PLAYING)
+      {
+        // Buffer underrun occurred, resume playback
+        alSourcePlay(uiSource);
+        err = alGetError();
+        if (err != 0)
+        {
+          ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
+        }
+      }
+    }
+    else
+    {
+      soundSyncEvent.Wait();
+    }
+  }
 }
 
-#endif //HAVE_OPENAL
-
+#endif  // HAVE_OPENAL