Remove unnecessary Src/ folders

1 files changed, 0 insertions, 331 deletions

diff --git a/Source/Core/AudioCommon/Src/OpenALStream.cpp b/Source/Core/AudioCommon/Src/OpenALStream.cpp
deleted file mode 100644
index ce59dc6c57..0000000000
--- a/Source/Core/AudioCommon/Src/OpenALStream.cpp
+++ /dev/null
@@ -1,331 +0,0 @@
-// Copyright 2013 Dolphin Emulator Project
-// Licensed under GPLv2
-// Refer to the license.txt file included.
-
-#include "aldlist.h"
-#include "OpenALStream.h"
-#include "DPL2Decoder.h"
-
-#if defined HAVE_OPENAL && HAVE_OPENAL
-
-soundtouch::SoundTouch soundTouch;
-
-//
-// AyuanX: Spec says OpenAL1.1 is thread safe already
-//
-bool OpenALStream::Start()
-{
-	ALDeviceList *pDeviceList = NULL;
-	ALCcontext *pContext = NULL;
-	ALCdevice *pDevice = NULL;
-	bool bReturn = false;
-
-	pDeviceList = new ALDeviceList();
-	if ((pDeviceList) && (pDeviceList->GetNumDevices()))
-	{
-		char *defDevName = pDeviceList->GetDeviceName(pDeviceList->GetDefaultDevice());
-
-		WARN_LOG(AUDIO, "Found OpenAL device %s", defDevName);
-
-		pDevice = alcOpenDevice(defDevName);
-		if (pDevice)
-		{
-			pContext = alcCreateContext(pDevice, NULL);
-			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(std::mem_fun(&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);
-		}
-		delete pDeviceList;
-	}
-	else
-	{
-		PanicAlertT("OpenAL: can't find sound devices");
-	}
-
-	// Initialize DPL2 parameters
-	dpl2reset();
-
-	soundTouch.clear();
-	return bReturn;
-}
-
-void OpenALStream::Stop()
-{
-	threadData = 1;
-	// kick the thread if it's waiting
-	soundSyncEvent.Set();
-
-	soundTouch.clear();
-
-	thread.join();
-
-	alSourceStop(uiSource);
-	alSourcei(uiSource, AL_BUFFER, 0);
-
-	// Clean up buffers and sources
-	alDeleteSources(1, &uiSource);
-	uiSource = 0;
-	alDeleteBuffers(numBuffers, uiBuffers);
-
-	ALCcontext *pContext = alcGetCurrentContext();
-	ALCdevice *pDevice = alcGetContextsDevice(pContext);
-
-	alcMakeContextCurrent(NULL);
-	alcDestroyContext(pContext);
-	alcCloseDevice(pDevice);
-}
-
-void OpenALStream::SetVolume(int volume)
-{
-	fVolume = (float)volume / 100.0f;
-
-	if (uiSource)
-		alSourcef(uiSource, AL_GAIN, fVolume);
-}
-
-void OpenALStream::Update()
-{
-	soundSyncEvent.Set();
-}
-
-void OpenALStream::Clear(bool mute)
-{
-	m_muted = mute;
-
-	if(m_muted)
-	{
-		soundTouch.clear();
-		alSourceStop(uiSource);
-	}
-	else
-	{
-		alSourcePlay(uiSource);
-	}
-}
-
-void OpenALStream::SoundLoop()
-{
-	Common::SetCurrentThreadName("Audio thread - openal");
-
-	bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder;
-#if defined(__APPLE__)
-	bool float32_capable = false;
-	const ALenum AL_FORMAT_STEREO_FLOAT32 = 0;
-	// OSX does not have the alext AL_FORMAT_51CHN32 yet.
-	surround_capable = false;
-	const ALenum AL_FORMAT_51CHN32 = 0;
-#else
-	bool float32_capable = true;
-#endif
-
-	u32 ulFrequency = m_mixer->GetSampleRate();
-	numBuffers = Core::g_CoreStartupParameter.iLatency + 2; // OpenAL requires a minimum of two buffers
-
-	memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
-	uiSource = 0;
-
-	// Generate some AL Buffers for streaming
-	alGenBuffers(numBuffers, (ALuint *)uiBuffers);
-	// Generate a Source to playback the Buffers
-	alGenSources(1, &uiSource);
-
-	// Short Silence
-	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 (surround_capable)
-			alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * FRAME_SURROUND_FLOAT, 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 (!threadData)
-	{
-		// 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);
-
-		// 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 << 16);
-
-		soundTouch.putSamples(dest, numSamples);
-
-		if (iBuffersProcessed == iBuffersFilled)
-		{
-			alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed);
-			iBuffersFilled = 0;
-		}
-
-		if (iBuffersProcessed)
-		{
-			float rate = m_mixer->GetCurrentSpeed();
-			if (rate <= 0)
-			{
-				Core::RequestRefreshInfo();
-				rate = 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)
-			{
-				// Adjust SETTING_SEQUENCE_MS to balance between lag vs hollow audio
-				soundTouch.setSetting(SETTING_SEQUENCE_MS, (int)(1 / (rate * rate)));
-				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);
-				alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * FRAME_SURROUND_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;
-				}
-				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 << 16));
-
-					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
-