Reformat all the things. Have fun with merge conflicts.

Source/Android/jni/ButtonManager.h

@@ -191,22 +191,22 @@ namespace ButtonManager
 {
 private:
   ButtonState m_state;
+
 public:
   Button() : m_state(BUTTON_RELEASED) {}
   void SetState(ButtonState state) { m_state = state; }
   bool Pressed() { return m_state == BUTTON_PRESSED; }
-
   ~Button() {}
 };
 class Axis
 {
 private:
   float m_value;
+
 public:
   Axis() : m_value(0.0f) {}
   void SetValue(float value) { m_value = value; }
   float AxisValue() { return m_value; }
-
   ~Axis() {}
 };
 
@@ -219,10 +219,10 @@ namespace ButtonManager
   const float _neg;
   sBind(int padID, ButtonType buttontype, BindType bindtype, int bind, float neg)
       : _padID(padID), _buttontype(buttontype), _bindtype(bindtype), _bind(bind), _neg(neg)
-		{}
+  {
+  }
 };
 
-
 class InputDevice
 {
 private:
@@ -232,9 +232,9 @@ namespace ButtonManager
 
   // Key is padID and ButtonType
   std::map<std::pair<int, ButtonType>, sBind*> _inputbinds;
+
 public:
-		InputDevice(std::string dev)
+  InputDevice(std::string dev) : _dev(dev) {}
-			: _dev(dev) {}
   ~InputDevice()
   {
     for (const auto& bind : _inputbinds)

Source/Core/AudioCommon/AOSoundStream.cpp

@@ -6,8 +6,8 @@
 
 #include "AudioCommon/AOSoundStream.h"
 #include "AudioCommon/Mixer.h"
-#include "Common/MsgHandler.h"
 #include "Common/Logging/Log.h"
+#include "Common/MsgHandler.h"
 
 #if defined(HAVE_AO) && HAVE_AO
 

Source/Core/AudioCommon/AOSoundStream.h

@@ -38,10 +38,6 @@ public:
   void Stop() override;
   void Update() override;
 
-	static bool isValid()
+  static bool isValid() { return true; }
-	{
-		return true;
-	}
-
 #endif
 };

Source/Core/AudioCommon/AlsaSoundStream.cpp

@@ -6,13 +6,12 @@
 
 #include "AudioCommon/AlsaSoundStream.h"
 #include "Common/CommonTypes.h"
-#include "Common/Thread.h"
 #include "Common/Logging/Log.h"
+#include "Common/Thread.h"
 
 AlsaSound::AlsaSound()
-	: m_thread_status(ALSAThreadStatus::STOPPED)
+    : m_thread_status(ALSAThreadStatus::STOPPED), handle(nullptr),
-	, handle(nullptr)
+      frames_to_deliver(FRAME_COUNT_MIN)
-	, frames_to_deliver(FRAME_COUNT_MIN)
 {
 }
 
@@ -79,7 +78,6 @@ void AlsaSound::SoundLoop()
   m_thread_status.store(ALSAThreadStatus::STOPPED);
 }
 
-
 void AlsaSound::Clear(bool muted)
 {
   m_muted = muted;
@@ -188,7 +186,9 @@ bool AlsaSound::AlsaInit()
   // it is probably a bad idea to try to send more than one buffer of data
   if ((unsigned int)frames_to_deliver > buffer_size)
     frames_to_deliver = buffer_size;
-	NOTICE_LOG(AUDIO, "ALSA gave us a %ld sample \"hardware\" buffer with %d periods. Will send %d samples per fragments.\n", buffer_size, periods, frames_to_deliver);
+  NOTICE_LOG(AUDIO, "ALSA gave us a %ld sample \"hardware\" buffer with %d periods. Will send %d "
+                    "samples per fragments.\n",
+             buffer_size, periods, frames_to_deliver);
 
   snd_pcm_sw_params_alloca(&swparams);
 
@@ -232,4 +232,3 @@ void AlsaSound::AlsaShutdown()
     handle = nullptr;
   }
 }
-

Source/Core/AudioCommon/AlsaSoundStream.h

@@ -28,11 +28,7 @@ public:
   void Update() override;
   void Clear(bool) override;
 
-	static bool isValid()
+  static bool isValid() { return true; }
-	{
-		return true;
-	}
-
 private:
   // maximum number of frames the buffer can hold
   static constexpr size_t BUFFER_SIZE_MAX = 8192;

Source/Core/AudioCommon/AudioCommon.cpp

@@ -2,22 +2,21 @@
 // Licensed under GPLv2+
 // Refer to the license.txt file included.
 
-
-#include "AudioCommon/AlsaSoundStream.h"
-#include "AudioCommon/AOSoundStream.h"
 #include "AudioCommon/AudioCommon.h"
+#include "AudioCommon/AOSoundStream.h"
+#include "AudioCommon/AlsaSoundStream.h"
 #include "AudioCommon/CoreAudioSoundStream.h"
 #include "AudioCommon/Mixer.h"
 #include "AudioCommon/NullSoundStream.h"
 #include "AudioCommon/OpenALStream.h"
 #include "AudioCommon/OpenSLESStream.h"
 #include "AudioCommon/PulseAudioStream.h"
-#include "AudioCommon/XAudio2_7Stream.h"
 #include "AudioCommon/XAudio2Stream.h"
+#include "AudioCommon/XAudio2_7Stream.h"
 #include "Common/Common.h"
 #include "Common/FileUtil.h"
-#include "Common/MsgHandler.h"
 #include "Common/Logging/Log.h"
+#include "Common/MsgHandler.h"
 #include "Core/ConfigManager.h"
 #include "Core/Movie.h"
 
@@ -58,8 +57,8 @@ namespace AudioCommon
 
   if (!g_sound_stream && NullSound::isValid())
   {
-			WARN_LOG(AUDIO, "Could not initialize backend %s, using %s instead.",
+    WARN_LOG(AUDIO, "Could not initialize backend %s, using %s instead.", backend.c_str(),
-				backend.c_str(), BACKEND_NULLSOUND);
+             BACKEND_NULLSOUND);
     g_sound_stream = new NullSound();
   }
 
@@ -68,8 +67,8 @@ namespace AudioCommon
     UpdateSoundStream();
     if (!g_sound_stream->Start())
     {
-				ERROR_LOG(AUDIO, "Could not start backend %s, using %s instead",
+      ERROR_LOG(AUDIO, "Could not start backend %s, using %s instead", backend.c_str(),
-					  backend.c_str(), BACKEND_NULLSOUND);
+                BACKEND_NULLSOUND);
       delete g_sound_stream;
       g_sound_stream = new NullSound();
       g_sound_stream->Start();

Source/Core/AudioCommon/AudioCommon.h

@@ -7,7 +7,6 @@
 #include "AudioCommon/SoundStream.h"
 #include "Common/CommonTypes.h"
 
-
 class CMixer;
 
 extern SoundStream* g_sound_stream;

Source/Core/AudioCommon/CoreAudioSoundStream.cpp

@@ -7,15 +7,13 @@
 #include "AudioCommon/CoreAudioSoundStream.h"
 #include "Common/Logging/Log.h"
 
-OSStatus CoreAudioSound::callback(void *inRefCon,
+OSStatus CoreAudioSound::callback(void* inRefCon, AudioUnitRenderActionFlags* ioActionFlags,
-	AudioUnitRenderActionFlags *ioActionFlags,
                                   const AudioTimeStamp* inTimeStamp, UInt32 inBusNumber,
                                   UInt32 inNumberFrames, AudioBufferList* ioData)
 {
   for (UInt32 i = 0; i < ioData->mNumberBuffers; i++)
-		((CoreAudioSound *)inRefCon)->m_mixer->
+    ((CoreAudioSound*)inRefCon)
-			Mix((short *)ioData->mBuffers[i].mData,
+        ->m_mixer->Mix((short*)ioData->mBuffers[i].mData, ioData->mBuffers[i].mDataByteSize / 4);
-				ioData->mBuffers[i].mDataByteSize / 4);
 
   return noErr;
 }
@@ -47,12 +45,9 @@ bool CoreAudioSound::Start()
     return false;
   }
 
-	FillOutASBDForLPCM(format, m_mixer->GetSampleRate(),
+  FillOutASBDForLPCM(format, m_mixer->GetSampleRate(), 2, 16, 16, false, false, false);
-				2, 16, 16, false, false, false);
+  err = AudioUnitSetProperty(audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0,
-	err = AudioUnitSetProperty(audioUnit,
+                             &format, sizeof(AudioStreamBasicDescription));
-				kAudioUnitProperty_StreamFormat,
-				kAudioUnitScope_Input, 0, &format,
-				sizeof(AudioStreamBasicDescription));
   if (err != noErr)
   {
     ERROR_LOG(AUDIO, "error setting audio format");
@@ -61,19 +56,15 @@ bool CoreAudioSound::Start()
 
   callback_struct.inputProc = callback;
   callback_struct.inputProcRefCon = this;
-	err = AudioUnitSetProperty(audioUnit,
+  err = AudioUnitSetProperty(audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input,
-				kAudioUnitProperty_SetRenderCallback,
+                             0, &callback_struct, sizeof callback_struct);
-				kAudioUnitScope_Input, 0, &callback_struct,
-				sizeof callback_struct);
   if (err != noErr)
   {
     ERROR_LOG(AUDIO, "error setting audio callback");
     return false;
   }
 
-	err = AudioUnitSetParameter(audioUnit,
+  err = AudioUnitSetParameter(audioUnit, kHALOutputParam_Volume, kAudioUnitScope_Output, 0,
-					kHALOutputParam_Volume,
-					kAudioUnitScope_Output, 0,
                               m_volume / 100., 0);
   if (err != noErr)
     ERROR_LOG(AUDIO, "error setting volume");
@@ -100,9 +91,7 @@ void CoreAudioSound::SetVolume(int volume)
   OSStatus err;
   m_volume = volume;
 
-	err = AudioUnitSetParameter(audioUnit,
+  err = AudioUnitSetParameter(audioUnit, kHALOutputParam_Volume, kAudioUnitScope_Output, 0,
-					kHALOutputParam_Volume,
-					kAudioUnitScope_Output, 0,
                               volume / 100., 0);
   if (err != noErr)
     ERROR_LOG(AUDIO, "error setting volume");

Source/Core/AudioCommon/CoreAudioSoundStream.h

@@ -20,19 +20,13 @@ public:
   void Stop() override;
   void Update() override;
 
-	static bool isValid()
+  static bool isValid() { return true; }
-	{
-		return true;
-	}
-
 private:
   AudioUnit audioUnit;
   int m_volume;
 
-	static OSStatus callback(void *inRefCon,
+  static OSStatus callback(void* inRefCon, AudioUnitRenderActionFlags* ioActionFlags,
-		AudioUnitRenderActionFlags *ioActionFlags,
+                           const AudioTimeStamp* inTimeStamp, UInt32 inBusNumber,
-		const AudioTimeStamp *inTimeStamp,
+                           UInt32 inNumberFrames, AudioBufferList* ioData);
-		UInt32 inBusNumber, UInt32 inNumberFrames,
-		AudioBufferList *ioData);
 #endif
 };

Source/Core/AudioCommon/DPL2Decoder.cpp

@@ -68,7 +68,8 @@ static T FIRFilter(const T *buf, int pos, int len, int count, const float *coeff
   if (pos >= count)
   {
     pos -= count;
-		count1 = count; count2 = 0;
+    count1 = count;
+    count2 = 0;
   }
   else
   {
@@ -80,7 +81,8 @@ static T FIRFilter(const T *buf, int pos, int len, int count, const float *coeff
   // high part of window
   const T* ptr = &buf[pos];
 
-	float r1 = DotProduct(count1, ptr, coefficients); coefficients += count1;
+  float r1 = DotProduct(count1, ptr, coefficients);
+  coefficients += count1;
   float r2 = DotProduct(count2, buf, coefficients);
   return T(r1 + r2);
 }
@@ -167,7 +169,6 @@ static float* DesignFIR(unsigned int *n, float* fc, float opt)
     g += 2 * w[end - i - 1];  // Total gain in filter
   }
 
-
   // Normalize gain
   g = 1 / g;
   for (u32 i = 0; i < *n; i++)
@@ -219,26 +220,24 @@ static float* CalculateCoefficients125HzLowpass(int rate)
 
 static float PassiveLock(float x)
 {
-	static const float MATAGCLOCK = 0.2f;  /* AGC range (around 1) where the matrix behaves passively */
+  static const float MATAGCLOCK =
+      0.2f; /* AGC range (around 1) where the matrix behaves passively */
   const float x1 = x - 1;
   const float ax1s = fabs(x - 1) * (1.0f / MATAGCLOCK);
   return x1 - x1 / (1 + ax1s * ax1s) + 1;
 }
 
-static void MatrixDecode(const float *in, const int k, const int il,
+static void MatrixDecode(const float* in, const int k, const int il, const int ir, bool decode_rear,
-	const int ir, bool decode_rear,
+                         const int _dlbuflen, float _l_fwr, float _r_fwr, float _lpr_fwr,
-	const int _dlbuflen,
+                         float _lmr_fwr, float* _adapt_l_gain, float* _adapt_r_gain,
-	float _l_fwr, float _r_fwr,
+                         float* _adapt_lpr_gain, float* _adapt_lmr_gain, float* _lf, float* _rf,
-	float _lpr_fwr, float _lmr_fwr,
+                         float* _lr, float* _rr, float* _cf)
-	float *_adapt_l_gain, float *_adapt_r_gain,
-	float *_adapt_lpr_gain, float *_adapt_lmr_gain,
-	float *_lf, float *_rf, float *_lr,
-	float *_rr, float *_cf)
 {
   static const float M9_03DB = 0.3535533906f;
   static const float MATAGCTRIG = 8.0f;  /* (Fuzzy) AGC trigger */
   static const float MATAGCDECAY = 1.0f; /* AGC baseline decay rate (1/samp.) */
-	static const float MATCOMPGAIN = 0.37f; /* Cross talk compensation gain,  0.50 - 0.55 is full cancellation. */
+  static const float MATCOMPGAIN =
+      0.37f; /* Cross talk compensation gain,  0.50 - 0.55 is full cancellation. */
 
   const int kr = (k + olddelay) % _dlbuflen;
   float l_gain = (_l_fwr + _r_fwr) / (1 + _l_fwr + _l_fwr);
@@ -357,12 +356,9 @@ void DPL2Decode(float *samples, int numsamples, float *out)
     /* Matrix encoded 2 channel sources */
     fwrbuf_l[k] = in[0];
     fwrbuf_r[k] = in[1];
-		MatrixDecode(in, k, 0, 1, true, dlbuflen,
+    MatrixDecode(in, k, 0, 1, true, dlbuflen, l_fwr, r_fwr, lpr_fwr, lmr_fwr, &adapt_l_gain,
-			l_fwr, r_fwr,
+                 &adapt_r_gain, &adapt_lpr_gain, &adapt_lmr_gain, &lf[0], &rf[0], &lr[0], &rr[0],
-			lpr_fwr, lmr_fwr,
+                 &cf[0]);
-			&adapt_l_gain, &adapt_r_gain,
-			&adapt_lpr_gain, &adapt_lmr_gain,
-			&lf[0], &rf[0], &lr[0], &rr[0], &cf[0]);
 
     out[cur + 0] = lf[k];
     out[cur + 1] = rf[k];

Source/Core/AudioCommon/Mixer.cpp

@@ -8,16 +8,15 @@
 #include "AudioCommon/Mixer.h"
 #include "Common/CommonFuncs.h"
 #include "Common/CommonTypes.h"
-#include "Common/MathUtil.h"
 #include "Common/Logging/Log.h"
+#include "Common/MathUtil.h"
 #include "Core/ConfigManager.h"
 
 #if _M_SSE >= 0x301 && !(defined __GNUC__ && !defined __SSSE3__)
 #include <tmmintrin.h>
 #endif
 
-CMixer::CMixer(unsigned int BackendSampleRate)
+CMixer::CMixer(unsigned int BackendSampleRate) : m_sampleRate(BackendSampleRate)
-	: m_sampleRate(BackendSampleRate)
 {
   INFO_LOG(AUDIO_INTERFACE, "Mixer is initialized");
 }
@@ -27,7 +26,8 @@ CMixer::~CMixer()
 }
 
 // Executed from sound stream thread
-unsigned int CMixer::MixerFifo::Mix(short* samples, unsigned int numSamples, bool consider_framelimit)
+unsigned int CMixer::MixerFifo::Mix(short* samples, unsigned int numSamples,
+                                    bool consider_framelimit)
 {
   unsigned int currentSample = 0;
 
@@ -47,8 +47,10 @@ unsigned int CMixer::MixerFifo::Mix(short* samples, unsigned int numSamples, boo
   float numLeft = (float)(((indexW - indexR) & INDEX_MASK) / 2);
   m_numLeftI = (numLeft + m_numLeftI * (CONTROL_AVG - 1)) / CONTROL_AVG;
   float offset = (m_numLeftI - low_waterwark) * CONTROL_FACTOR;
-	if (offset > MAX_FREQ_SHIFT) offset = MAX_FREQ_SHIFT;
+  if (offset > MAX_FREQ_SHIFT)
-	if (offset < -MAX_FREQ_SHIFT) offset = -MAX_FREQ_SHIFT;
+    offset = MAX_FREQ_SHIFT;
+  if (offset < -MAX_FREQ_SHIFT)
+    offset = -MAX_FREQ_SHIFT;
 
   // render numleft sample pairs to samples[]
   // advance indexR with sample position
@@ -167,7 +169,8 @@ void CMixer::PushStreamingSamples(const short *samples, unsigned int num_samples
     m_wave_writer_dtk.AddStereoSamplesBE(samples, num_samples);
 }
 
-void CMixer::PushWiimoteSpeakerSamples(const short *samples, unsigned int num_samples, unsigned int sample_rate)
+void CMixer::PushWiimoteSpeakerSamples(const short* samples, unsigned int num_samples,
+                                       unsigned int sample_rate)
 {
   short samples_stereo[MAX_SAMPLES * 2];
 

Source/Core/AudioCommon/Mixer.h

@@ -22,9 +22,9 @@ public:
   // Called from main thread
   void PushSamples(const short* samples, unsigned int num_samples);
   void PushStreamingSamples(const short* samples, unsigned int num_samples);
-	void PushWiimoteSpeakerSamples(const short* samples, unsigned int num_samples, unsigned int sample_rate);
+  void PushWiimoteSpeakerSamples(const short* samples, unsigned int num_samples,
+                                 unsigned int sample_rate);
   unsigned int GetSampleRate() const { return m_sampleRate; }
-
   void SetDMAInputSampleRate(unsigned int rate);
   void SetStreamInputSampleRate(unsigned int rate);
   void SetStreamingVolume(unsigned int lvolume, unsigned int rvolume);
@@ -38,7 +38,6 @@ public:
 
   float GetCurrentSpeed() const { return m_speed.load(); }
   void UpdateSpeed(float val) { m_speed.store(val); }
-
 private:
   static constexpr u32 MAX_SAMPLES = 1024 * 4;  // 128 ms
   static constexpr u32 INDEX_MASK = MAX_SAMPLES * 2 - 1;
@@ -50,14 +49,14 @@ private:
   {
   public:
     MixerFifo(CMixer* mixer, unsigned sample_rate)
-			: m_mixer(mixer)
+        : m_mixer(mixer), m_input_sample_rate(sample_rate)
-			, m_input_sample_rate(sample_rate)
     {
     }
     void PushSamples(const short* samples, unsigned int num_samples);
     unsigned int Mix(short* samples, unsigned int numSamples, bool consider_framelimit = true);
     void SetInputSampleRate(unsigned int rate);
     void SetVolume(unsigned int lvolume, unsigned int rvolume);
+
   private:
     CMixer* m_mixer;
     unsigned m_input_sample_rate;

Source/Core/AudioCommon/NullSoundStream.cpp

@@ -25,9 +25,12 @@ void NullSound::Update()
   // num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
   constexpr u32 stereo_16_bit_size = 4;
   constexpr u32 dma_length = 32;
-	const u64 audio_dma_period = SystemTimers::GetTicksPerSecond() / (AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length);
+  const u64 audio_dma_period =
+      SystemTimers::GetTicksPerSecond() /
+      (AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length);
   const u64 ais_samples_per_second = 48000 * stereo_16_bit_size;
-	const u64 num_samples_to_render = (audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond();
+  const u64 num_samples_to_render =
+      (audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond();
 
   m_mixer->Mix(m_realtime_buffer.data(), (unsigned int)num_samples_to_render);
 }

Source/Core/AudioCommon/NullSoundStream.h

@@ -18,7 +18,6 @@ public:
   void Update() override;
 
   static bool isValid() { return true; }
-
 private:
   static constexpr size_t BUFFER_SIZE = 48000 * 4 / 32;
 

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
@@ -149,7 +149,8 @@ void OpenALStream::SoundLoop()
   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.
+  // 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;
 
@@ -171,13 +172,16 @@ void OpenALStream::SoundLoop()
     if (surround_capable)
     {
       if (float32_capable)
-				alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * FRAME_SURROUND_FLOAT, ulFrequency);
+        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);
+        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);
+      alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * FRAME_STEREO_SHORT,
+                   ulFrequency);
     }
   }
   alSourceQueueBuffers(uiSource, numBuffers, uiBuffers);
@@ -209,11 +213,14 @@ void OpenALStream::SoundLoop()
     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 audio_dma_period = SystemTimers::GetTicksPerSecond() /
-		u64 num_samples_to_render = (audio_dma_period * ais_samples_per_second) / 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)
+    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);
@@ -256,7 +263,8 @@ void OpenALStream::SoundLoop()
       if (nSamples <= minSamples)
         continue;
 
-			// Remove the Buffer from the Queue.  (uiBuffer contains the Buffer ID for the unqueued Buffer)
+      // Remove the Buffer from the Queue.  (uiBuffer contains the Buffer ID for the unqueued
+      // Buffer)
       if (iBuffersFilled == 0)
       {
         alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
@@ -283,7 +291,8 @@ void OpenALStream::SoundLoop()
 
         if (float32_capable)
         {
-					alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
+          alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2,
+                       nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
         }
         else
         {
@@ -291,14 +300,16 @@ void OpenALStream::SoundLoop()
           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);
+          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.");
+          WARN_LOG(AUDIO,
+                   "Unable to set 5.1 surround mode.  Updating OpenAL Soft might fix this issue.");
           surround_capable = false;
         }
         else if (err != 0)
@@ -311,7 +322,8 @@ void OpenALStream::SoundLoop()
       {
         if (float32_capable)
         {
-					alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, nSamples * FRAME_STEREO_FLOAT, ulFrequency);
+          alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
+                       nSamples * FRAME_STEREO_FLOAT, ulFrequency);
           ALenum err = alGetError();
           if (err == AL_INVALID_ENUM)
           {
@@ -330,7 +342,8 @@ void OpenALStream::SoundLoop()
           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);
+          alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo,
+                       nSamples * FRAME_STEREO_SHORT, ulFrequency);
         }
       }
 
@@ -372,4 +385,3 @@ void OpenALStream::SoundLoop()
 }
 
 #endif  // HAVE_OPENAL
-

Source/Core/AudioCommon/OpenALStream.h

@@ -32,8 +32,8 @@
 #define BOOL SoundTouch_BOOL
 #endif
 
-#include <soundtouch/SoundTouch.h>
 #include <soundtouch/STTypes.h>
+#include <soundtouch/SoundTouch.h>
 
 #ifdef __APPLE__
 #undef BOOL
@@ -57,10 +57,7 @@ class OpenALStream final : public SoundStream
 {
 #if defined HAVE_OPENAL && HAVE_OPENAL
 public:
-	OpenALStream() : uiSource(0)
+  OpenALStream() : uiSource(0) {}
-	{
-	}
-
   bool Start() override;
   void SoundLoop() override;
   void SetVolume(int volume) override;
@@ -69,7 +66,6 @@ public:
   void Update() override;
 
   static bool isValid() { return true; }
-
 private:
   std::thread thread;
   std::atomic<bool> m_run_thread;

Source/Core/AudioCommon/OpenSLESStream.cpp

@@ -39,7 +39,8 @@ static void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
 
   // Render to the fresh buffer
   g_mixer->Mix(reinterpret_cast<short*>(buffer[curBuffer]), BUFFER_SIZE_IN_SAMPLES);
-	SLresult result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, buffer[curBuffer], sizeof(buffer[0]));
+  SLresult result =
+      (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, buffer[curBuffer], sizeof(buffer[0]));
   curBuffer ^= 1;  // Switch buffer
 
   // Comment from sample code:
@@ -64,15 +65,13 @@ bool OpenSLESStream::Start()
   assert(SL_RESULT_SUCCESS == result);
 
   SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 2};
-	SLDataFormat_PCM format_pcm = {
+  SLDataFormat_PCM format_pcm = {SL_DATAFORMAT_PCM,
-		SL_DATAFORMAT_PCM,
                                  2,
                                  m_mixer->GetSampleRate() * 1000,
                                  SL_PCMSAMPLEFORMAT_FIXED_16,
                                  SL_PCMSAMPLEFORMAT_FIXED_16,
                                  SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT,
-		SL_BYTEORDER_LITTLEENDIAN
+                                 SL_BYTEORDER_LITTLEENDIAN};
-	};
 
   SLDataSource audioSrc = {&loc_bufq, &format_pcm};
 
@@ -83,15 +82,17 @@ bool OpenSLESStream::Start()
   // create audio player
   const SLInterfaceID ids[2] = {SL_IID_BUFFERQUEUE, SL_IID_VOLUME};
   const SLboolean req[2] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
-	result = (*engineEngine)->CreateAudioPlayer(engineEngine, &bqPlayerObject, &audioSrc, &audioSnk, 2, ids, req);
+  result =
+      (*engineEngine)
+          ->CreateAudioPlayer(engineEngine, &bqPlayerObject, &audioSrc, &audioSnk, 2, ids, req);
   assert(SL_RESULT_SUCCESS == result);
 
   result = (*bqPlayerObject)->Realize(bqPlayerObject, SL_BOOLEAN_FALSE);
   assert(SL_RESULT_SUCCESS == result);
   result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_PLAY, &bqPlayerPlay);
   assert(SL_RESULT_SUCCESS == result);
-	result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_BUFFERQUEUE,
+  result =
-		&bqPlayerBufferQueue);
+      (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_BUFFERQUEUE, &bqPlayerBufferQueue);
   assert(SL_RESULT_SUCCESS == result);
   result = (*bqPlayerBufferQueue)->RegisterCallback(bqPlayerBufferQueue, bqPlayerCallback, nullptr);
   assert(SL_RESULT_SUCCESS == result);

Source/Core/AudioCommon/OpenSLESStream.h

@@ -16,7 +16,6 @@ public:
   bool Start() override;
   void Stop() override;
   static bool isValid() { return true; }
-
 private:
   std::thread thread;
   Common::Event soundSyncEvent;