UnrealEngineUWP/Engine/Source/Runtime/Android/AndroidAudio/Private/AndroidAudioSource.cpp

// Copyright 1998-2018 Epic Games, Inc. All Rights Reserved.

/*------------------------------------------------------------------------------------
	FSLESSoundSource.
------------------------------------------------------------------------------------*/

#include "AndroidAudioDevice.h"
#include "AudioDecompress.h"
#include "ContentStreaming.h"

// Keep track of OpenSLES track counts, if they exceed 12, then log the error but prevent new sounds
static int32 GTrackCount = 0;

// Track when the last realize error ocurred, if they 
static double GLastRealizeErrorTimeSec = 0.0;

// How much time is needed to pass since last error before a new sound is allowed to play
int32 GCVarAndroidRealizeErrorWaitThresholdMs = 35;
TAutoConsoleVariable<int32> CVarAndroidRealizeErrorWaitThresholdMs(TEXT("au.AndroidRealizeErrorWaitThresholdMs"), GCVarAndroidRealizeErrorWaitThresholdMs, TEXT("Sets number of ms to wait before allowing new player request after realize buffer error (default: 35)"), ECVF_Default);



// Callback that is registered if the source needs to loop
void OpenSLBufferQueueCallback( SLAndroidSimpleBufferQueueItf InQueueInterface, void* pContext ) 
{
	FSLESSoundSource* SoundSource = (FSLESSoundSource*)pContext;
	if( SoundSource )
	{
		SoundSource->OnRequeueBufferCallback( InQueueInterface );
	}
}

// Requeues buffer to loop Sound Source
void FSLESSoundSource::OnRequeueBufferCallback( SLAndroidSimpleBufferQueueItf InQueueInterface ) 
{
	if (!bStreamedSound)
	{
		SLresult result = (*SL_PlayerBufferQueue)->Enqueue(SL_PlayerBufferQueue, SLESBuffer->AudioData, SLESBuffer->GetSize() );
		if(result != SL_RESULT_SUCCESS) 
		{ 
			UE_LOG( LogAndroidAudio, Warning, TEXT("FAILED OPENSL BUFFER Enqueue SL_PlayerBufferQueue (Requeing)"));  
		}
		bHasLooped = true;
	}
	else
	{
		// Enqueue the previously decoded buffer
		if (RealtimeAsyncTask)
		{
			RealtimeAsyncTask->EnsureCompletion();
			switch(RealtimeAsyncTask->GetTask().GetTaskType())
			{
			case ERealtimeAudioTaskType::Decompress:
				bHasLooped = RealtimeAsyncTask->GetTask().GetBufferLooped();
				break;

			case ERealtimeAudioTaskType::Procedural:
				AudioBuffers[BufferInUse].AudioDataSize = RealtimeAsyncTask->GetTask().GetBytesWritten();
				break;
			}

			delete RealtimeAsyncTask;
			RealtimeAsyncTask = nullptr;
		}

		// Sound decoding is complete, just waiting to finish playing
		if (bBuffersToFlush)
		{
			// set the player's state to stopped
			SLresult result = (*SL_PlayerPlayInterface)->SetPlayState(SL_PlayerPlayInterface, SL_PLAYSTATE_STOPPED);
			check(SL_RESULT_SUCCESS == result);

			return;
		}

		SLresult result = (*SL_PlayerBufferQueue)->Enqueue(SL_PlayerBufferQueue, AudioBuffers[BufferInUse].AudioData, AudioBuffers[BufferInUse].AudioDataSize );
		if(result != SL_RESULT_SUCCESS) 
		{ 
			UE_LOG( LogAndroidAudio, Warning, TEXT("FAILED OPENSL BUFFER Enqueue SL_PlayerBufferQueue (Requeing)"));  
		}

		// Switch to the next buffer and decode for the next time the callback fires if we didn't just get the last buffer
		BufferInUse = !BufferInUse;
		if (bHasLooped == false || WaveInstance->LoopingMode != LOOP_Never)
		{
			// Do this in the callback thread instead of creating an asynchronous task (thread id from callback is not consistent and use of TLS for stats causes issues)
			if (ReadMorePCMData(BufferInUse, EDataReadMode::Synchronous))
			{
				// If this is a synchronous source we may get notified immediately that we have looped
				bHasLooped = true;
			}
		}
	}
}

bool FSLESSoundSource::CreatePlayer()
{
	if (GTrackCount >= 12)
	{
		UE_LOG(LogAndroidAudio, Warning, TEXT("Too many audio tracks to create new player! (%d)"), GTrackCount);
		return false;
	}

	const double CurrentTimeSec = FPlatformTime::Seconds();
	const double LastTimeSinceErrorMs = (CurrentTimeSec - GLastRealizeErrorTimeSec) * 1000.0;
	if (LastTimeSinceErrorMs < (double)GCVarAndroidRealizeErrorWaitThresholdMs)
	{
		UE_LOG(LogAndroidAudio, Warning, TEXT("Last time since Realize Error: %f ms"), LastTimeSinceErrorMs);
		return false;
	}

	// data info
	SLDataLocator_AndroidSimpleBufferQueue LocationBuffer	= {		SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 1 };
		
	// PCM Info
	SLDataFormat_PCM PCM_Format				= {		SL_DATAFORMAT_PCM, SLuint32(SLESBuffer->NumChannels), SLuint32( SLESBuffer->SampleRate * 1000 ),	
		SL_PCMSAMPLEFORMAT_FIXED_16, SL_PCMSAMPLEFORMAT_FIXED_16, 
		SLESBuffer->NumChannels == 2 ? ( SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT ) : SL_SPEAKER_FRONT_CENTER, 
		SL_BYTEORDER_LITTLEENDIAN };
		
	SLDataSource SoundDataSource			= {		&LocationBuffer, &PCM_Format};
		
	// configure audio sink
	SLDataLocator_OutputMix Output_Mix		= {		SL_DATALOCATOR_OUTPUTMIX, ((FSLESAudioDevice *)AudioDevice)->SL_OutputMixObject};
	SLDataSink AudioSink					= {		&Output_Mix, NULL};


	// create audio player
	const SLInterfaceID	ids[] = {SL_IID_BUFFERQUEUE, SL_IID_VOLUME};
	const SLboolean		req[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
	SLresult result;

	{
		SCOPE_CYCLE_COUNTER(STAT_AudioAndroidSourcePlayerCreateTime);

		result = (*Device->SL_EngineEngine)->CreateAudioPlayer(Device->SL_EngineEngine, &SL_PlayerObject, &SoundDataSource, &AudioSink, sizeof(ids) / sizeof(SLInterfaceID), ids, req);
		if (result != SL_RESULT_SUCCESS)
		{
			UE_LOG(LogAndroidAudio, Warning, TEXT("FAILED OPENSL BUFFER CreateAudioPlayer 0x%x"), result);
			return false;
		}

	}
	
	bool bFailedSetup = false;
	
	// realize the player
	{
		SCOPE_CYCLE_COUNTER(STAT_AudioAndroidSourcePlayerRealize);

		result = (*SL_PlayerObject)->Realize(SL_PlayerObject, SL_BOOLEAN_FALSE);

		++GTrackCount;

		if (result != SL_RESULT_SUCCESS)
		{
			GLastRealizeErrorTimeSec = FPlatformTime::Seconds();
			UE_LOG(LogAndroidAudio, Warning, TEXT("FAILED OPENSL BUFFER Realize 0x%x. NumChannels: %d, SampleRate: %d, TrackCount: %d"), result, SLESBuffer->NumChannels, SLESBuffer->SampleRate, GTrackCount);
			return false;
		}
	}
		
	// get the play interface
	result = (*SL_PlayerObject)->GetInterface(SL_PlayerObject, SL_IID_PLAY, &SL_PlayerPlayInterface);
	if (result != SL_RESULT_SUCCESS) { UE_LOG(LogAndroidAudio, Warning, TEXT("FAILED OPENSL BUFFER GetInterface SL_IID_PLAY 0x%x"), result); bFailedSetup |= true; }
	// volume
	result = (*SL_PlayerObject)->GetInterface(SL_PlayerObject, SL_IID_VOLUME, &SL_VolumeInterface);
	if (result != SL_RESULT_SUCCESS) { UE_LOG(LogAndroidAudio, Warning, TEXT("FAILED OPENSL BUFFER GetInterface SL_IID_VOLUME 0x%x"), result); bFailedSetup |= true; }
	// buffer system
	result = (*SL_PlayerObject)->GetInterface(SL_PlayerObject, SL_IID_BUFFERQUEUE, &SL_PlayerBufferQueue);
	if (result != SL_RESULT_SUCCESS) { UE_LOG(LogAndroidAudio, Warning, TEXT("FAILED OPENSL BUFFER GetInterface SL_IID_BUFFERQUEUE 0x%x"), result); bFailedSetup |= true; }
	
	return bFailedSetup == false;
}

void FSLESSoundSource::DestroyPlayer()
{
	if( SL_PlayerObject )
	{
		// close it down...
		(*SL_PlayerObject)->Destroy(SL_PlayerObject);			
		SL_PlayerObject			= NULL;
		SL_PlayerPlayInterface	= NULL;
		SL_PlayerBufferQueue	= NULL;
		SL_VolumeInterface		= NULL;

		--GTrackCount;
	}
}

bool FSLESSoundSource::EnqueuePCMBuffer( bool bLoop)
{
	SLresult result;
	// If looping, register a callback to requeue the buffer
	if( bLoop ) 
	{
		result = (*SL_PlayerBufferQueue)->RegisterCallback(SL_PlayerBufferQueue, OpenSLBufferQueueCallback, (void*)this);
		if (result != SL_RESULT_SUCCESS) { UE_LOG(LogAndroidAudio, Warning, TEXT("FAILED OPENSL BUFFER QUEUE RegisterCallback 0x%x "), result); return false; }
	}
	
	result = (*SL_PlayerBufferQueue)->Enqueue(SL_PlayerBufferQueue, SLESBuffer->AudioData, SLESBuffer->GetSize() );
	if (result != SL_RESULT_SUCCESS) {
		UE_LOG(LogAndroidAudio, Warning, TEXT("FAILED OPENSL BUFFER Enqueue SL_PlayerBufferQueue 0x%x params( %p, %d)"), result, SLESBuffer->AudioData, int32(SLESBuffer->GetSize()));
		if (bLoop)
		{
			result = (*SL_PlayerBufferQueue)->RegisterCallback(SL_PlayerBufferQueue, NULL, NULL);
		}
		return false;
	}

	bStreamedSound = false;
	bHasLooped = false;
	bHasPositionUpdated = false;
	bBuffersToFlush = false;

	return true;
}

bool FSLESSoundSource::ReadMorePCMData(const int32 BufferIndex, EDataReadMode DataReadMode)
{
	USoundWave* WaveData = WaveInstance->WaveData;
	if (WaveData && WaveData->bProcedural)
	{
		const int32 MaxSamples = BufferSize / sizeof(int16);
		if (DataReadMode == EDataReadMode::Synchronous || WaveData->bCanProcessAsync == false)
		{
			const int32 BytesWritten = WaveData->GeneratePCMData(AudioBuffers[BufferIndex].AudioData, MaxSamples);
			AudioBuffers[BufferIndex].AudioDataSize = BytesWritten;
		}
		else
		{
			RealtimeAsyncTask = new FAsyncRealtimeAudioTask(WaveData, AudioBuffers[BufferIndex].AudioData, MaxSamples);
			RealtimeAsyncTask->StartBackgroundTask();
		}

		// we're never actually "looping" here.
		return false;
	}
	else
	{
		if (DataReadMode == EDataReadMode::Synchronous)
		{
			return SLESBuffer->ReadCompressedData(AudioBuffers[BufferIndex].AudioData, WaveInstance->LoopingMode != LOOP_Never);
		}
		else
		{
			RealtimeAsyncTask = new FAsyncRealtimeAudioTask(SLESBuffer, AudioBuffers[BufferIndex].AudioData, WaveInstance->LoopingMode != LOOP_Never, DataReadMode == EDataReadMode::AsynchronousSkipFirstFrame);
			RealtimeAsyncTask->StartBackgroundTask();
			return false;
		}
	}
}


bool FSLESSoundSource::EnqueuePCMRTBuffer( bool bLoop )
{
	if (AudioBuffers[0].AudioData || AudioBuffers[1].AudioData)
	{
		UE_LOG( LogAndroidAudio, Warning, TEXT("Enqueue PCMRT with buffers already allocated"));
	}
	FMemory::Memzero( AudioBuffers, sizeof( SLESAudioBuffer ) * 2 );

	// Set up double buffer area to decompress to
	BufferSize = SLESBuffer->GetRTBufferSize() * SLESBuffer->NumChannels;

	AudioBuffers[0].AudioData = (uint8*)FMemory::Malloc(BufferSize);
	AudioBuffers[0].AudioDataSize = BufferSize;

	AudioBuffers[1].AudioData = (uint8*)FMemory::Malloc(BufferSize);
	AudioBuffers[1].AudioDataSize = BufferSize;

	// Only use the cached data if we're starting from the beginning, otherwise we'll have to take a synchronous hit
	if (WaveInstance->WaveData && WaveInstance->WaveData->CachedRealtimeFirstBuffer && WaveInstance->StartTime == 0.f)
	{
		FMemory::Memcpy((uint8*)AudioBuffers[0].AudioData, WaveInstance->WaveData->CachedRealtimeFirstBuffer, BufferSize);
		ReadMorePCMData(1, EDataReadMode::AsynchronousSkipFirstFrame);
	}
	else
	{
		UE_LOG(LogAndroidAudio, Warning, TEXT("Performing synchronous decode on audio thread with '%s'. This may cause hitching on the game thread if performed too often."), *WaveInstance->WaveData->GetName());

		ReadMorePCMData(0, EDataReadMode::Synchronous);
		ReadMorePCMData(1, EDataReadMode::Asynchronous);
	}

	SLresult result;

	// callback is used to submit and decompress next buffer
	result = (*SL_PlayerBufferQueue)->RegisterCallback(SL_PlayerBufferQueue, OpenSLBufferQueueCallback, (void*)this);
	
	// queue one sound buffer, as that is all Android will accept
	if(result == SL_RESULT_SUCCESS) 
	{
		result = (*SL_PlayerBufferQueue)->Enqueue(SL_PlayerBufferQueue, AudioBuffers[0].AudioData, AudioBuffers[0].AudioDataSize );
		if (result != SL_RESULT_SUCCESS) { UE_LOG(LogAndroidAudio, Warning, TEXT("FAILED OPENSL BUFFER Enqueue SL_PlayerBufferQueue 0x%x params( %p, %d)"), result, SLESBuffer->AudioData, int32(SLESBuffer->GetSize())); return false; }
	}
	else
	{
		return false;
	}

	bStreamedSound = true;
	bHasLooped = false;
	bBuffersToFlush = false;
	bHasPositionUpdated = false;
	BufferInUse = 1;
	return true;
}

/**
 * Initializes a source with a given wave instance and prepares it for playback.
 *
 * @param	WaveInstance	wave instance being primed for playback
 * @return	TRUE if initialization was successful, FALSE otherwise
 */
bool FSLESSoundSource::Init( FWaveInstance* InWaveInstance )
{
	FSoundSource::InitCommon();

	// don't do anything if no volume! THIS APPEARS TO HAVE THE VOLUME IN TIME, CHECK HERE THOUGH IF ISSUES
	if( InWaveInstance && ( InWaveInstance->GetActualVolume()) <= 0 )
	{
		return false;
	}

	if (SLESBuffer && SLESBuffer->ResourceID == 0)
	{
		UE_LOG( LogAndroidAudio, Warning, TEXT(" InitSoundSouce with Buffer already allocated"));
		delete SLESBuffer;
		SLESBuffer = nullptr;
		Buffer = nullptr;
	}

	if (SL_PlayerObject)
	{
		UE_LOG( LogAndroidAudio, Warning, TEXT(" InitSoundSouce with PlayerObject not NULL, possible leak"));
	}
	
	// Find matching buffer.
	SLESBuffer = FSLESSoundBuffer::Init( (FSLESAudioDevice *)AudioDevice, InWaveInstance->WaveData );
	Buffer = SLESBuffer;

	if( SLESBuffer && InWaveInstance->WaveData->NumChannels <= 2 && InWaveInstance->WaveData->GetSampleRateForCurrentPlatform() <= 48000 )
	{
		SCOPE_CYCLE_COUNTER( STAT_AudioSourceInitTime );
		
		bool bFailedSetup = false;

		if (CreatePlayer())
		{
			WaveInstance = InWaveInstance;

			if (WaveInstance->StartTime > 0.f)
			{
				SLESBuffer->Seek(WaveInstance->StartTime);
			}

			switch( SLESBuffer->Format)
			{
				case SoundFormat_PCM:
					bFailedSetup |= !EnqueuePCMBuffer( InWaveInstance->LoopingMode != LOOP_Never );
					break;
				case SoundFormat_PCMRT:
				case SoundFormat_Streaming:
					bFailedSetup |= !EnqueuePCMRTBuffer( InWaveInstance->LoopingMode != LOOP_Never );
					break;
				default:
					bFailedSetup = true;
			}
		}
		else
		{
			bFailedSetup = true;
		}
		
		// clean up the madness if anything we need failed
		if( bFailedSetup )
		{
			UE_LOG( LogAndroidAudio, Warning, TEXT(" Setup failed %s"), *InWaveInstance->WaveData->GetName());
			DestroyPlayer();
			return false;
		}
		
		Update();
		
		// Initialization was successful.
		return true;
	}
	else
	{
		// Failed to initialize source.
		// These occurences appear to potentially lead to leaks
		UE_LOG( LogAndroidAudio, Warning, TEXT("Init SoundSource failed on %s"), *InWaveInstance->WaveData->GetName());
		UE_LOG( LogAndroidAudio, Warning, TEXT("  SampleRate %d"), InWaveInstance->WaveData->GetSampleRateForCurrentPlatform());
		UE_LOG( LogAndroidAudio, Warning, TEXT("  Channels %d"), InWaveInstance->WaveData->NumChannels);

		if (SLESBuffer && SLESBuffer->ResourceID == 0)
		{
			delete SLESBuffer;
			SLESBuffer = nullptr;
			Buffer = nullptr;
		} 
	}
	return false;
}


FSLESSoundSource::FSLESSoundSource( class FAudioDevice* InAudioDevice )
	:	FSoundSource( InAudioDevice ),
		Device((FSLESAudioDevice *)InAudioDevice),
		SLESBuffer( NULL ),
		bStreamedSound(false),
		bBuffersToFlush(false),
		BufferSize(0),
		BufferInUse(0),
		VolumePreviousUpdate(-1.0f),
		bHasLooped(false),
		bHasPositionUpdated(false),
		SL_PlayerObject(NULL),
		SL_PlayerPlayInterface(NULL),
		SL_PlayerBufferQueue(NULL),
		SL_VolumeInterface(NULL),
		RealtimeAsyncTask(NULL)
{
	Buffer = NULL;
	FMemory::Memzero( AudioBuffers, sizeof( AudioBuffers ) );
}

/**
 * Clean up any hardware referenced by the sound source
 */
FSLESSoundSource::~FSLESSoundSource( void )
{
	DestroyPlayer();

	ReleaseResources();
}

void FSLESSoundSource::ReleaseResources()
{
	if (RealtimeAsyncTask)
	{
		RealtimeAsyncTask->EnsureCompletion();
		delete RealtimeAsyncTask;
		RealtimeAsyncTask = nullptr;
	}

	FMemory::Free( AudioBuffers[0].AudioData);
	FMemory::Free( AudioBuffers[1].AudioData);

	FMemory::Memzero( AudioBuffers, sizeof( AudioBuffers ) );

	if (SLESBuffer && SLESBuffer->ResourceID == 0)
	{
		delete SLESBuffer;
	}
	SLESBuffer = nullptr;
	Buffer = nullptr;
}

/**
 * Updates the source specific parameter like e.g. volume and pitch based on the associated
 * wave instance.	
 */
void FSLESSoundSource::Update( void )
{
	SCOPE_CYCLE_COUNTER( STAT_AudioUpdateSources );
	
	if( !WaveInstance || Paused )
	{
		return;
	}

	FSoundSource::UpdateCommon();
	
	float Volume = WaveInstance->GetActualVolume();
	if (SetStereoBleed())
	{
		// Emulate the bleed to rear speakers followed by stereo fold down
		Volume *= 1.25f;
	}
	Volume *= AudioDevice->GetPlatformAudioHeadroom();
	Volume = FMath::Clamp(Volume, 0.0f, MAX_VOLUME);
	
	Volume = FSoundSource::GetDebugVolume(Volume);

	// Set whether to apply reverb
	SetReverbApplied(true);

	SetFilterFrequency();
	
	// Avoid doing the log calculation each update by only doing it if the volume changed
	if (Volume != VolumePreviousUpdate)
	{
		VolumePreviousUpdate = Volume;
		static const int64 MinVolumeMillibel = -12000;
		if (Volume > 0.0f)
		{
			// Convert volume to millibels.
			SLmillibel MaxMillibel = 0;
			(*SL_VolumeInterface)->GetMaxVolumeLevel(SL_VolumeInterface, &MaxMillibel);
			SLmillibel VolumeMillibel = (SLmillibel)FMath::Clamp<int64>((int64)(2000.0f * log10f(Volume)), MinVolumeMillibel, (int64)MaxMillibel);
			SLresult result = (*SL_VolumeInterface)->SetVolumeLevel(SL_VolumeInterface, VolumeMillibel);
			check(SL_RESULT_SUCCESS == result);
		}
		else
		{
			SLresult result = (*SL_VolumeInterface)->SetVolumeLevel(SL_VolumeInterface, MinVolumeMillibel);
			check(SL_RESULT_SUCCESS == result);
		}
	}

}

/**
 * Plays the current wave instance.	
 */
void FSLESSoundSource::Play( void )
{
	if( WaveInstance )
	{
		// Reset the previous volume on play so it can be set at least once in the update function
		VolumePreviousUpdate = -1.0f;

		// Update volume now before starting play
		Paused = false;
		Update();

		// set the player's state to playing
		SLresult result = (*SL_PlayerPlayInterface)->SetPlayState(SL_PlayerPlayInterface, SL_PLAYSTATE_PLAYING);
		check(SL_RESULT_SUCCESS == result);

		Playing = true;
	}
}

/**
 * Stops the current wave instance and detaches it from the source.	
 */
void FSLESSoundSource::Stop( void )
{
	IStreamingManager::Get().GetAudioStreamingManager().RemoveStreamingSoundSource(this);

	if (SL_PlayerPlayInterface)
	{
		// set the player's state to stopped
		SLresult result = (*SL_PlayerPlayInterface)->SetPlayState(SL_PlayerPlayInterface, SL_PLAYSTATE_STOPPED);
		check(SL_RESULT_SUCCESS == result);
	}

	if (WaveInstance)
	{
		// Unregister looping callback
		if (WaveInstance->LoopingMode != LOOP_Never)
		{
			SLresult result = (*SL_PlayerBufferQueue)->RegisterCallback(SL_PlayerBufferQueue, NULL, NULL);
			check(SL_RESULT_SUCCESS == result);
		}
	}
	DestroyPlayer();
	ReleaseResources();
	Paused = false;
	Playing = false;
	SLESBuffer = nullptr;
	Buffer = nullptr;

	DestroyPlayer();
	ReleaseResources();

	Paused = false;
	Playing = false;
	SLESBuffer = nullptr;
	Buffer = nullptr;

	FSoundSource::Stop();
}

/**
 * Pauses playback of current wave instance.
 */
void FSLESSoundSource::Pause( void )
{
	if( WaveInstance )
	{
		Paused = true;
		
		// set the player's state to paused
		SLresult result = (*SL_PlayerPlayInterface)->SetPlayState(SL_PlayerPlayInterface, SL_PLAYSTATE_PAUSED);
		check(SL_RESULT_SUCCESS == result);
		
	}
}

/** 
 * Returns TRUE if the source has finished playing
 */
bool FSLESSoundSource::IsSourceFinished( void )
{
	SLuint32 PlayState;
	
	// set the player's state to playing
	SLresult result = (*SL_PlayerPlayInterface)->GetPlayState(SL_PlayerPlayInterface, &PlayState);
	check(SL_RESULT_SUCCESS == result);
	
	if( PlayState == SL_PLAYSTATE_STOPPED )
	{
		return true;
	}

	if (WaveInstance && WaveInstance->LoopingMode == LOOP_Never)
	{
		// if it wasn't that simple, see if we're at the end position
		SLmillisecond PositionMs;
		SLmillisecond DurationMs;

		result = (*SL_PlayerPlayInterface)->GetPosition(SL_PlayerPlayInterface, &PositionMs);
		check(SL_RESULT_SUCCESS == result);

		result = (*SL_PlayerPlayInterface)->GetDuration(SL_PlayerPlayInterface, &DurationMs);
		check(SL_RESULT_SUCCESS == result);

		// on some android devices, the value for GetPosition wraps back to 0 when the playback is done, however it's very possible
		// for us to try to check for IsSourceFinished when the Position is genuinely "0". Therefore, we'll flip bHasPositionUpdated once
		// we've actually started the sound to denote a wrap-back 0 position versus a real 0 position
		if ((DurationMs != SL_TIME_UNKNOWN && PositionMs == DurationMs) || (PositionMs == 0 && bHasPositionUpdated))
		{
			return true;
		}
		else if (!bHasPositionUpdated && PositionMs > 0)
		{
			bHasPositionUpdated = true;
		}
	}
	

	return false;
}

/**
 * Queries the status of the currently associated wave instance.
 *
 * @return	TRUE if the wave instance/ source has finished playback and FALSE if it is 
 *			currently playing or paused.
 */
bool FSLESSoundSource::IsFinished( void )
{
	if( WaveInstance )
	{
		// Check for a non starved, stopped source
		if( IsSourceFinished() )
		{
			// Notify the wave instance that it has finished playing
			WaveInstance->NotifyFinished();
			return true;
		}
		else 
		{
			if (bHasLooped)
			{
				switch (WaveInstance->LoopingMode)
				{
					case LOOP_Forever:
						bHasLooped = false;
						break;

					case LOOP_Never:
						bBuffersToFlush = true;
						break;

					case LOOP_WithNotification:
						bHasLooped = false;
						// Notify the wave instance that it has finished playing.
						WaveInstance->NotifyFinished();
						break;
				};
			}
		}
		
		return false;
	}
	return true;
}