Reformat all the things. Have fun with merge conflicts.

1 files changed, 191 insertions, 192 deletions

diff --git a/Source/Core/AudioCommon/AlsaSoundStream.cpp b/Source/Core/AudioCommon/AlsaSoundStream.cpp
index 19a21acf66..3598958c75 100644
--- a/Source/Core/AudioCommon/AlsaSoundStream.cpp
+++ b/Source/Core/AudioCommon/AlsaSoundStream.cpp
@@ -6,230 +6,229 @@
 
 #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)
-	, handle(nullptr)
-	, frames_to_deliver(FRAME_COUNT_MIN)
+    : m_thread_status(ALSAThreadStatus::STOPPED), handle(nullptr),
+      frames_to_deliver(FRAME_COUNT_MIN)
 {
 }
 
 bool AlsaSound::Start()
 {
-	m_thread_status.store(ALSAThreadStatus::RUNNING);
-	if (!AlsaInit())
-	{
-		m_thread_status.store(ALSAThreadStatus::STOPPED);
-		return false;
-	}
-
-	thread = std::thread(&AlsaSound::SoundLoop, this);
-	return true;
+  m_thread_status.store(ALSAThreadStatus::RUNNING);
+  if (!AlsaInit())
+  {
+    m_thread_status.store(ALSAThreadStatus::STOPPED);
+    return false;
+  }
+
+  thread = std::thread(&AlsaSound::SoundLoop, this);
+  return true;
 }
 
 void AlsaSound::Stop()
 {
-	m_thread_status.store(ALSAThreadStatus::STOPPING);
+  m_thread_status.store(ALSAThreadStatus::STOPPING);
 
-	//Give the opportunity to the audio thread
-	//to realize we are stopping the emulation
-	cv.notify_one();
-	thread.join();
+  // Give the opportunity to the audio thread
+  // to realize we are stopping the emulation
+  cv.notify_one();
+  thread.join();
 }
 
 void AlsaSound::Update()
 {
-	// don't need to do anything here.
+  // don't need to do anything here.
 }
 
 // Called on audio thread.
 void AlsaSound::SoundLoop()
 {
-	Common::SetCurrentThreadName("Audio thread - alsa");
-	while (m_thread_status.load() != ALSAThreadStatus::STOPPING)
-	{
-		while (m_thread_status.load() == ALSAThreadStatus::RUNNING)
-		{
-			m_mixer->Mix(mix_buffer, frames_to_deliver);
-			int rc = snd_pcm_writei(handle, mix_buffer, frames_to_deliver);
-			if (rc == -EPIPE)
-			{
-				// Underrun
-				snd_pcm_prepare(handle);
-			}
-			else if (rc < 0)
-			{
-				ERROR_LOG(AUDIO, "writei fail: %s", snd_strerror(rc));
-			}
-		}
-		if (m_thread_status.load() == ALSAThreadStatus::PAUSED)
-		{
-			snd_pcm_drop(handle); // Stop sound output
-
-			// Block until thread status changes.
-			std::unique_lock<std::mutex> lock(cv_m);
-			cv.wait(lock, [this]{ return m_thread_status.load() != ALSAThreadStatus::PAUSED; });
-
-			snd_pcm_prepare(handle); // resume sound output
-		}
-	}
-	AlsaShutdown();
-	m_thread_status.store(ALSAThreadStatus::STOPPED);
+  Common::SetCurrentThreadName("Audio thread - alsa");
+  while (m_thread_status.load() != ALSAThreadStatus::STOPPING)
+  {
+    while (m_thread_status.load() == ALSAThreadStatus::RUNNING)
+    {
+      m_mixer->Mix(mix_buffer, frames_to_deliver);
+      int rc = snd_pcm_writei(handle, mix_buffer, frames_to_deliver);
+      if (rc == -EPIPE)
+      {
+        // Underrun
+        snd_pcm_prepare(handle);
+      }
+      else if (rc < 0)
+      {
+        ERROR_LOG(AUDIO, "writei fail: %s", snd_strerror(rc));
+      }
+    }
+    if (m_thread_status.load() == ALSAThreadStatus::PAUSED)
+    {
+      snd_pcm_drop(handle);  // Stop sound output
+
+      // Block until thread status changes.
+      std::unique_lock<std::mutex> lock(cv_m);
+      cv.wait(lock, [this] { return m_thread_status.load() != ALSAThreadStatus::PAUSED; });
+
+      snd_pcm_prepare(handle);  // resume sound output
+    }
+  }
+  AlsaShutdown();
+  m_thread_status.store(ALSAThreadStatus::STOPPED);
 }
 
-
 void AlsaSound::Clear(bool muted)
 {
-	m_muted = muted;
-	m_thread_status.store(muted ? ALSAThreadStatus::PAUSED : ALSAThreadStatus::RUNNING);
-	cv.notify_one(); // Notify thread that status has changed
+  m_muted = muted;
+  m_thread_status.store(muted ? ALSAThreadStatus::PAUSED : ALSAThreadStatus::RUNNING);
+  cv.notify_one();  // Notify thread that status has changed
 }
 
 bool AlsaSound::AlsaInit()
 {
-	unsigned int sample_rate = m_mixer->GetSampleRate();
-	int err;
-	int dir;
-	snd_pcm_sw_params_t *swparams;
-	snd_pcm_hw_params_t *hwparams;
-	snd_pcm_uframes_t buffer_size,buffer_size_max;
-	unsigned int periods;
-
-	err = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "Audio open error: %s\n", snd_strerror(err));
-		return false;
-	}
-
-	snd_pcm_hw_params_alloca(&hwparams);
-
-	err = snd_pcm_hw_params_any(handle, hwparams);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "Broken configuration for this PCM: %s\n", snd_strerror(err));
-		return false;
-	}
-
-	err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "Access type not available: %s\n", snd_strerror(err));
-		return false;
-	}
-
-	err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_LE);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "Sample format not available: %s\n", snd_strerror(err));
-		return false;
-	}
-
-	dir = 0;
-	err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &sample_rate, &dir);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "Rate not available: %s\n", snd_strerror(err));
-		return false;
-	}
-
-	err = snd_pcm_hw_params_set_channels(handle, hwparams, CHANNEL_COUNT);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "Channels count not available: %s\n", snd_strerror(err));
-		return false;
-	}
-
-	periods = BUFFER_SIZE_MAX / FRAME_COUNT_MIN;
-	err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &periods, &dir);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "Cannot set maximum periods per buffer: %s\n", snd_strerror(err));
-		return false;
-	}
-
-	buffer_size_max = BUFFER_SIZE_MAX;
-	err = snd_pcm_hw_params_set_buffer_size_max(handle, hwparams, &buffer_size_max);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "Cannot set maximum buffer size: %s\n", snd_strerror(err));
-		return false;
-	}
-
-	err = snd_pcm_hw_params(handle, hwparams);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "Unable to install hw params: %s\n", snd_strerror(err));
-		return false;
-	}
-
-	err = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "Cannot get buffer size: %s\n", snd_strerror(err));
-		return false;
-	}
-
-	err = snd_pcm_hw_params_get_periods_max(hwparams, &periods, &dir);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "Cannot get periods: %s\n", snd_strerror(err));
-		return false;
-	}
-
-	//periods is the number of fragments alsa can wait for during one
-	//buffer_size
-	frames_to_deliver = buffer_size / periods;
-	//limit the minimum size. pulseaudio advertises a minimum of 32 samples.
-	if (frames_to_deliver < FRAME_COUNT_MIN)
-		frames_to_deliver = FRAME_COUNT_MIN;
-	//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);
-
-	snd_pcm_sw_params_alloca(&swparams);
-
-	err = snd_pcm_sw_params_current(handle, swparams);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "cannot init sw params: %s\n", snd_strerror(err));
-		return false;
-	}
-
-	err = snd_pcm_sw_params_set_start_threshold(handle, swparams, 0U);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "cannot set start thresh: %s\n", snd_strerror(err));
-		return false;
-	}
-
-	err = snd_pcm_sw_params(handle, swparams);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "cannot set sw params: %s\n", snd_strerror(err));
-		return false;
-	}
-
-	err = snd_pcm_prepare(handle);
-	if (err < 0)
-	{
-		ERROR_LOG(AUDIO, "Unable to prepare: %s\n", snd_strerror(err));
-		return false;
-	}
-	NOTICE_LOG(AUDIO, "ALSA successfully initialized.\n");
-	return true;
+  unsigned int sample_rate = m_mixer->GetSampleRate();
+  int err;
+  int dir;
+  snd_pcm_sw_params_t* swparams;
+  snd_pcm_hw_params_t* hwparams;
+  snd_pcm_uframes_t buffer_size, buffer_size_max;
+  unsigned int periods;
+
+  err = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "Audio open error: %s\n", snd_strerror(err));
+    return false;
+  }
+
+  snd_pcm_hw_params_alloca(&hwparams);
+
+  err = snd_pcm_hw_params_any(handle, hwparams);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "Broken configuration for this PCM: %s\n", snd_strerror(err));
+    return false;
+  }
+
+  err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "Access type not available: %s\n", snd_strerror(err));
+    return false;
+  }
+
+  err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_LE);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "Sample format not available: %s\n", snd_strerror(err));
+    return false;
+  }
+
+  dir = 0;
+  err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &sample_rate, &dir);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "Rate not available: %s\n", snd_strerror(err));
+    return false;
+  }
+
+  err = snd_pcm_hw_params_set_channels(handle, hwparams, CHANNEL_COUNT);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "Channels count not available: %s\n", snd_strerror(err));
+    return false;
+  }
+
+  periods = BUFFER_SIZE_MAX / FRAME_COUNT_MIN;
+  err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &periods, &dir);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "Cannot set maximum periods per buffer: %s\n", snd_strerror(err));
+    return false;
+  }
+
+  buffer_size_max = BUFFER_SIZE_MAX;
+  err = snd_pcm_hw_params_set_buffer_size_max(handle, hwparams, &buffer_size_max);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "Cannot set maximum buffer size: %s\n", snd_strerror(err));
+    return false;
+  }
+
+  err = snd_pcm_hw_params(handle, hwparams);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "Unable to install hw params: %s\n", snd_strerror(err));
+    return false;
+  }
+
+  err = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "Cannot get buffer size: %s\n", snd_strerror(err));
+    return false;
+  }
+
+  err = snd_pcm_hw_params_get_periods_max(hwparams, &periods, &dir);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "Cannot get periods: %s\n", snd_strerror(err));
+    return false;
+  }
+
+  // periods is the number of fragments alsa can wait for during one
+  // buffer_size
+  frames_to_deliver = buffer_size / periods;
+  // limit the minimum size. pulseaudio advertises a minimum of 32 samples.
+  if (frames_to_deliver < FRAME_COUNT_MIN)
+    frames_to_deliver = FRAME_COUNT_MIN;
+  // 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);
+
+  snd_pcm_sw_params_alloca(&swparams);
+
+  err = snd_pcm_sw_params_current(handle, swparams);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "cannot init sw params: %s\n", snd_strerror(err));
+    return false;
+  }
+
+  err = snd_pcm_sw_params_set_start_threshold(handle, swparams, 0U);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "cannot set start thresh: %s\n", snd_strerror(err));
+    return false;
+  }
+
+  err = snd_pcm_sw_params(handle, swparams);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "cannot set sw params: %s\n", snd_strerror(err));
+    return false;
+  }
+
+  err = snd_pcm_prepare(handle);
+  if (err < 0)
+  {
+    ERROR_LOG(AUDIO, "Unable to prepare: %s\n", snd_strerror(err));
+    return false;
+  }
+  NOTICE_LOG(AUDIO, "ALSA successfully initialized.\n");
+  return true;
 }
 
 void AlsaSound::AlsaShutdown()
 {
-	if (handle != nullptr)
-	{
-		snd_pcm_drop(handle);
-		snd_pcm_close(handle);
-		handle = nullptr;
-	}
+  if (handle != nullptr)
+  {
+    snd_pcm_drop(handle);
+    snd_pcm_close(handle);
+    handle = nullptr;
+  }
 }
-