Merge branch 'immediate-removal' into GLSL-master

Conflicts:
	Source/Core/VideoCommon/Src/PixelShaderGen.cpp
	Source/Plugins/Plugin_VideoSoftware/Src/SWRenderer.cpp

immediate-removal is a new created branch seperated from master but reverted the revert of immediate-removal
so we get less conflicts by merging

1 files changed, 191 insertions, 34 deletions

diff --git a/Source/Core/AudioCommon/Src/OpenALStream.cpp b/Source/Core/AudioCommon/Src/OpenALStream.cpp
index 2abd357782..cfe6a40630 100644
--- a/Source/Core/AudioCommon/Src/OpenALStream.cpp
+++ b/Source/Core/AudioCommon/Src/OpenALStream.cpp
@@ -15,13 +15,14 @@
 // Official SVN repository and contact information can be found at
 // http://code.google.com/p/dolphin-emu/
 
-#include <functional>
-
 #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
 //
@@ -35,14 +36,21 @@ bool OpenALStream::Start()
 	pDeviceList = new ALDeviceList();
 	if ((pDeviceList) && (pDeviceList->GetNumDevices()))
 	{
-		char *defDevName = pDeviceList-> \
-			GetDeviceName(pDeviceList->GetDefaultDevice());
+		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;
@@ -50,8 +58,7 @@ bool OpenALStream::Start()
 			else
 			{
 				alcCloseDevice(pDevice);
-				PanicAlertT("OpenAL: can't create context "
-					"for device %s", defDevName);
+				PanicAlertT("OpenAL: can't create context for device %s", defDevName);
 			}
 		}
 		else
@@ -65,6 +72,10 @@ bool OpenALStream::Start()
 		PanicAlertT("OpenAL: can't find sound devices");
 	}
 
+	// Initialise DPL2 parameters
+	dpl2reset();
+
+	soundTouch.clear();
 	return bReturn;
 }
 
@@ -74,6 +85,8 @@ void OpenALStream::Stop()
 	// kick the thread if it's waiting
 	soundSyncEvent.Set();
 
+	soundTouch.clear();
+
 	thread.join();
 
 	alSourceStop(uiSource);
@@ -82,7 +95,7 @@ void OpenALStream::Stop()
 	// Clean up buffers and sources
 	alDeleteSources(1, &uiSource);
 	uiSource = 0;
-	alDeleteBuffers(OAL_NUM_BUFFERS, uiBuffers);
+	alDeleteBuffers(numBuffers, uiBuffers);
 
 	ALCcontext *pContext = alcGetCurrentContext();
 	ALCdevice *pDevice = alcGetContextsDevice(pContext);
@@ -111,6 +124,7 @@ void OpenALStream::Clear(bool mute)
 
 	if(m_muted)
 	{
+		soundTouch.clear();
 		alSourceStop(uiSource);
 	}
 	else
@@ -124,20 +138,29 @@ void OpenALStream::SoundLoop()
 	Common::SetCurrentThreadName("Audio thread - openal");
 
 	u32 ulFrequency = m_mixer->GetSampleRate();
+	numBuffers = Core::g_CoreStartupParameter.iLatency + 2; // OpenAL requires a minimum of two buffers
 
-	memset(uiBuffers, 0, OAL_NUM_BUFFERS * sizeof(ALuint));
+	memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
 	uiSource = 0;
 
 	// Generate some AL Buffers for streaming
-	alGenBuffers(OAL_NUM_BUFFERS, (ALuint *)uiBuffers);
+	alGenBuffers(numBuffers, (ALuint *)uiBuffers);
 	// Generate a Source to playback the Buffers
 	alGenSources(1, &uiSource);
 
 	// Short Silence
+	memset(sampleBuffer, 0, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * numBuffers);
 	memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * 4);
-	for (int i = 0; i < OAL_NUM_BUFFERS; i++)
-		alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, OAL_MAX_SAMPLES, ulFrequency);
-	alSourceQueueBuffers(uiSource, OAL_NUM_BUFFERS, uiBuffers);
+	for (int i = 0; i < numBuffers; i++)
+	{
+#if !defined(__APPLE__)
+		if (Core::g_CoreStartupParameter.bDPL2Decoder)
+			alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency);
+		else
+#endif
+			alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * 2 * 2, ulFrequency);
+	}
+	alSourceQueueBuffers(uiSource, numBuffers, uiBuffers);
 	alSourcePlay(uiSource);
 	
 	// Set the default sound volume as saved in the config file.
@@ -148,41 +171,175 @@ void OpenALStream::SoundLoop()
 
 	ALint iBuffersFilled = 0;
 	ALint iBuffersProcessed = 0;
-	ALuint uiBufferTemp[OAL_NUM_BUFFERS] = {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);
+
+	bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder;
+#if defined(__APPLE__)
+	bool float32_capable = false;
+#else
+	bool float32_capable = true;
+#endif
 
 	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;
+
+		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 * 2 * 2 * OAL_MAX_BUFFERS];
+		for (u32 i = 0; i < numSamples; ++i)
+		{
+			dest[i * 2 + 0] = (float)realtimeBuffer[i * 2 + 0] / (1 << 16);
+			dest[i * 2 + 1] = (float)realtimeBuffer[i * 2 + 1] / (1 << 16);
+		}
+
+		soundTouch.putSamples(dest, numSamples);
+
 		if (iBuffersProcessed == iBuffersFilled)
 		{
 			alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed);
 			iBuffersFilled = 0;
 		}
 
-		unsigned int numSamples = m_mixer->GetNumSamples();
-
-		if (iBuffersProcessed && (numSamples >= OAL_THRESHOLD))
+		if (iBuffersProcessed)
 		{
-			numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
-			// Remove the Buffer from the Queue.  (uiBuffer contains the Buffer ID for the unqueued Buffer)
-			if (iBuffersFilled == 0)
-				alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
-
-			m_mixer->Mix(realtimeBuffer, numSamples);
-			alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, realtimeBuffer, numSamples * 4, ulFrequency);
-			alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
-			iBuffersFilled++;
-
-			if (iBuffersFilled == OAL_NUM_BUFFERS)
-				alSourcePlay(uiSource);
+			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);
+			}
+			unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS);
+			if (nSamples > 0)
+			{
+				// 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 defined(__APPLE__)
+				// OSX does not have the alext AL_FORMAT_51CHN32 yet.
+				surround_capable = false;
+#else
+				if (surround_capable)
+				{
+					float dpl2[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
+					dpl2decode(sampleBuffer, nSamples, dpl2);
+
+					alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * SIZE_FLOAT * SURROUND_CHANNELS, 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);
+					}
+				}
+#endif
+				if (!surround_capable)
+				{
+#if !defined(__APPLE__)
+					if (float32_capable)
+					{
+						alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, nSamples * 4 * 2, 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);
+						}
+
+					}
+#endif
+					if (!float32_capable)
+					{
+						// Convert the samples from float to short
+						short stereo[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS];
+						for (u32 i = 0; i < nSamples; ++i)
+						{
+							stereo[i * 2 + 0] = (short)((float)sampleBuffer[i * 2 + 0] * (1 << 16));
+							stereo[i * 2 + 1] = (short)((float)sampleBuffer[i * 2 + 1] * (1 << 16));
+						}
+						alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo, nSamples * 2 * 2, 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);
+					ALenum 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);
+					ALenum err = alGetError();
+					if (err != 0)
+					{
+						ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
+					}
+				}
+			}
 		}
-		else if (numSamples >= OAL_THRESHOLD)
+		else
 		{
-			ALint state = 0;
-			alGetSourcei(uiSource, AL_SOURCE_STATE, &state);
-			if (state == AL_STOPPED)
-				alSourcePlay(uiSource);
+			soundSyncEvent.Wait();
 		}
-		soundSyncEvent.Wait();
 	}
 }