Remove unnecessary Src/ folders

1 files changed, 331 insertions, 0 deletions

diff --git a/Source/Core/AudioCommon/OpenALStream.cpp b/Source/Core/AudioCommon/OpenALStream.cpp
new file mode 100644
index 0000000000..ce59dc6c57
--- /dev/null
+++ b/Source/Core/AudioCommon/OpenALStream.cpp
@@ -0,0 +1,331 @@
+// 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
+