svn merge -r 129841:129910 svn+ssh://svn.lindenlab.com/svn/linden/branches/moss/pluginapi_05-merge@129910

svn merge -r 129913:131718 svn+ssh://svn.lindenlab.com/svn/linden/branches/pluginapi/pluginapi_05

Some branch shenannigans in the pluginapi_05 branch caused this to become a two-part merge.

21 files changed, 5631 insertions, 22 deletions

diff --git a/indra/llaudio/CMakeLists.txt b/indra/llaudio/CMakeLists.txt
index 235248ee73..80245fd569 100644
--- a/indra/llaudio/CMakeLists.txt
+++ b/indra/llaudio/CMakeLists.txt
@@ -4,15 +4,16 @@ project(llaudio)
 
 include(00-Common)
 include(Audio)
+include(LLAudio)
 include(FMOD)
 include(OPENAL)
 include(LLCommon)
 include(LLMath)
 include(LLMessage)
 include(LLVFS)
-include(LLMedia)
 
 include_directories(
+    ${LLAUDIO_INCLUDE_DIRS}
     ${FMOD_INCLUDE_DIR}
     ${LLCOMMON_INCLUDE_DIRS}
     ${LLMATH_INCLUDE_DIRS}
@@ -24,42 +25,43 @@ include_directories(
     ${VORBIS_INCLUDE_DIRS}
     ${OPENAL_LIB_INCLUDE_DIRS}
     ${FREEAULT_LIB_INCLUDE_DIRS}
-    ${LLMEDIA_INCLUDE_DIRS}
     )
 
 set(llaudio_SOURCE_FILES
-    audioengine.cpp
-    listener.cpp
+    llaudioengine.cpp
+    lllistener.cpp
     llaudiodecodemgr.cpp
-    vorbisdecode.cpp
-    vorbisencode.cpp
+    llvorbisdecode.cpp
+    llvorbisencode.cpp
     )
 
 set(llaudio_HEADER_FILES
     CMakeLists.txt
 
-    audioengine.h
-    listener.h
+    llaudioengine.h
+    lllistener.h
     llaudiodecodemgr.h
-    vorbisdecode.h
-    vorbisencode.h
-    windgen.h
+    llvorbisdecode.h
+    llvorbisencode.h
+    llwindgen.h
     )
 
 if (FMOD)
     list(APPEND llaudio_SOURCE_FILES
-         audioengine_fmod.cpp
-         listener_fmod.cpp
+         llaudioengine_fmod.cpp
+         lllistener_fmod.cpp
+         llstreamingaudio_fmod.cpp
          )
 
     list(APPEND llaudio_HEADER_FILES
-         audioengine_fmod.h
-         listener_fmod.h
+         llaudioengine_fmod.h
+         lllistener_fmod.h
+         llstreamingaudio_fmod.h
          )
 
     if (LINUX)
       if (${CXX_VERSION} MATCHES "4.[23]")
-        set_source_files_properties(audioengine_fmod.cpp
+        set_source_files_properties(llaudioengine_fmod.cpp
                                     COMPILE_FLAGS -Wno-error=write-strings)
       endif (${CXX_VERSION} MATCHES "4.[23]")
     endif (LINUX)
@@ -67,13 +69,13 @@ endif (FMOD)
 
 if (OPENAL)
   list(APPEND llaudio_SOURCE_FILES
-    audioengine_openal.cpp
-    listener_openal.cpp
+    llaudioengine_openal.cpp
+    lllistener_openal.cpp
     )
   
   list(APPEND llaudio_HEADER_FILES
-    audioengine_openal.h
-    listener_openal.h
+    llaudioengine_openal.h
+    lllistener_openal.h
     )
 endif (OPENAL)
 
diff --git a/indra/llaudio/llaudiodecodemgr.cpp b/indra/llaudio/llaudiodecodemgr.cpp
index 6a494d1983..099c4eba40 100644
--- a/indra/llaudio/llaudiodecodemgr.cpp
+++ b/indra/llaudio/llaudiodecodemgr.cpp
@@ -33,8 +33,8 @@
 
 #include "llaudiodecodemgr.h"
 
-#include "vorbisdecode.h"
-#include "audioengine.h"
+#include "llvorbisdecode.h"
+#include "llaudioengine.h"
 #include "lllfsthread.h"
 #include "llvfile.h"
 #include "llstring.h"
diff --git a/indra/llaudio/llaudioengine.cpp b/indra/llaudio/llaudioengine.cpp
new file mode 100644
index 0000000000..a28c94d00d
--- /dev/null
+++ b/indra/llaudio/llaudioengine.cpp
@@ -0,0 +1,1751 @@
+ /** 
+ * @file audioengine.cpp
+ * @brief implementation of LLAudioEngine class abstracting the Open
+ * AL audio support
+ *
+ * $LicenseInfo:firstyear=2000&license=viewergpl$
+ * 
+ * Copyright (c) 2000-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#include "linden_common.h"
+
+#include "llaudioengine.h"
+#include "llstreamingaudio.h"
+
+#include "llerror.h"
+#include "llmath.h"
+
+#include "sound_ids.h"  // temporary hack for min/max distances
+
+#include "llvfs.h"
+#include "lldir.h"
+#include "llaudiodecodemgr.h"
+#include "llassetstorage.h"
+
+
+// necessary for grabbing sounds from sim (implemented in viewer)	
+extern void request_sound(const LLUUID &sound_guid);
+
+LLAudioEngine* gAudiop = NULL;
+
+
+//
+// LLAudioEngine implementation
+//
+
+
+LLAudioEngine::LLAudioEngine()
+{
+	setDefaults();
+}
+
+
+LLAudioEngine::~LLAudioEngine()
+{
+}
+
+LLStreamingAudioInterface* LLAudioEngine::getStreamingAudioImpl()
+{
+	return mStreamingAudioImpl;
+}
+
+void LLAudioEngine::setStreamingAudioImpl(LLStreamingAudioInterface *impl)
+{
+	mStreamingAudioImpl = impl;
+}
+
+void LLAudioEngine::setDefaults()
+{
+	mMaxWindGain = 1.f;
+
+	mListenerp = NULL;
+
+	mMuted = false;
+	mUserData = NULL;
+
+	mLastStatus = 0;
+
+	mNumChannels = 0;
+	mEnableWind = false;
+
+	S32 i;
+	for (i = 0; i < MAX_CHANNELS; i++)
+	{
+		mChannels[i] = NULL;
+	}
+	for (i = 0; i < MAX_BUFFERS; i++)
+	{
+		mBuffers[i] = NULL;
+	}
+
+	mMasterGain = 1.f;
+	mNextWindUpdate = 0.f;
+
+	mStreamingAudioImpl = NULL;
+
+	for (U32 i = 0; i < LLAudioEngine::AUDIO_TYPE_COUNT; i++)
+		mSecondaryGain[i] = 1.0f;
+}
+
+
+bool LLAudioEngine::init(const S32 num_channels, void* userdata)
+{
+	setDefaults();
+
+	mNumChannels = num_channels;
+	mUserData = userdata;
+	
+	allocateListener();
+
+	// Initialize the decode manager
+	gAudioDecodeMgrp = new LLAudioDecodeMgr;
+
+	llinfos << "LLAudioEngine::init() AudioEngine successfully initialized" << llendl;
+
+	return true;
+}
+
+
+void LLAudioEngine::shutdown()
+{
+	// Clean up decode manager
+	delete gAudioDecodeMgrp;
+	gAudioDecodeMgrp = NULL;
+
+	// Clean up wind source
+	cleanupWind();
+
+	// Clean up audio sources
+	source_map::iterator iter_src;
+	for (iter_src = mAllSources.begin(); iter_src != mAllSources.end(); iter_src++)
+	{
+		delete iter_src->second;
+	}
+
+
+	// Clean up audio data
+	data_map::iterator iter_data;
+	for (iter_data = mAllData.begin(); iter_data != mAllData.end(); iter_data++)
+	{
+		delete iter_data->second;
+	}
+
+
+	// Clean up channels
+	S32 i;
+	for (i = 0; i < MAX_CHANNELS; i++)
+	{
+		delete mChannels[i];
+		mChannels[i] = NULL;
+	}
+
+	// Clean up buffers
+	for (i = 0; i < MAX_BUFFERS; i++)
+	{
+		delete mBuffers[i];
+		mBuffers[i] = NULL;
+	}
+}
+
+
+// virtual
+void LLAudioEngine::startInternetStream(const std::string& url)
+{
+	if (mStreamingAudioImpl)
+		mStreamingAudioImpl->start(url);
+}
+
+
+// virtual
+void LLAudioEngine::stopInternetStream()
+{
+	if (mStreamingAudioImpl)
+		mStreamingAudioImpl->stop();
+}
+
+// virtual
+void LLAudioEngine::pauseInternetStream(int pause)
+{
+	if (mStreamingAudioImpl)
+		mStreamingAudioImpl->pause(pause);
+}
+
+// virtual
+void LLAudioEngine::updateInternetStream()
+{
+	if (mStreamingAudioImpl)
+		mStreamingAudioImpl->update();
+}
+
+// virtual
+int LLAudioEngine::isInternetStreamPlaying()
+{
+	if (mStreamingAudioImpl)
+		return mStreamingAudioImpl->isPlaying();
+
+	return 0; // Stopped
+}
+
+
+// virtual
+void LLAudioEngine::setInternetStreamGain(F32 vol)
+{
+	if (mStreamingAudioImpl)
+		mStreamingAudioImpl->setGain(vol);
+}
+
+// virtual
+std::string LLAudioEngine::getInternetStreamURL()
+{
+	if (mStreamingAudioImpl)
+		return mStreamingAudioImpl->getURL();
+	else return std::string();
+}
+
+
+void LLAudioEngine::updateChannels()
+{
+	S32 i;
+	for (i = 0; i < MAX_CHANNELS; i++)
+	{
+		if (mChannels[i])
+		{
+			mChannels[i]->updateBuffer();
+			mChannels[i]->update3DPosition();
+			mChannels[i]->updateLoop();
+		}
+	}
+}
+
+static const F32 default_max_decode_time = .002f; // 2 ms
+void LLAudioEngine::idle(F32 max_decode_time)
+{
+	if (max_decode_time <= 0.f)
+	{
+		max_decode_time = default_max_decode_time;
+	}
+	
+	// "Update" all of our audio sources, clean up dead ones.
+	// Primarily does position updating, cleanup of unused audio sources.
+	// Also does regeneration of the current priority of each audio source.
+
+	if (getMuted())
+	{
+		setInternalGain(0.f);
+	}
+	else
+	{
+		setInternalGain(getMasterGain());
+	}
+
+	S32 i;
+	for (i = 0; i < MAX_BUFFERS; i++)
+	{
+		if (mBuffers[i])
+		{
+			mBuffers[i]->mInUse = false;
+		}
+	}
+
+	F32 max_priority = -1.f;
+	LLAudioSource *max_sourcep = NULL; // Maximum priority source without a channel
+	source_map::iterator iter;
+	for (iter = mAllSources.begin(); iter != mAllSources.end();)
+	{
+		LLAudioSource *sourcep = iter->second;
+
+		// Update this source
+		sourcep->update();
+		sourcep->updatePriority();
+
+		if (sourcep->isDone())
+		{
+			// The source is done playing, clean it up.
+			delete sourcep;
+			mAllSources.erase(iter++);
+			continue;
+		}
+
+		if (!sourcep->getChannel() && sourcep->getCurrentBuffer())
+		{
+			// We could potentially play this sound if its priority is high enough.
+			if (sourcep->getPriority() > max_priority)
+			{
+				max_priority = sourcep->getPriority();
+				max_sourcep = sourcep;
+			}
+		}
+
+		// Move on to the next source
+		iter++;
+	}
+
+	// Now, do priority-based organization of audio sources.
+	// All channels used, check priorities.
+	// Find channel with lowest priority
+	if (max_sourcep)
+	{
+		LLAudioChannel *channelp = getFreeChannel(max_priority);
+		if (channelp)
+		{
+			//llinfos << "Replacing source in channel due to priority!" << llendl;
+			max_sourcep->setChannel(channelp);
+			channelp->setSource(max_sourcep);
+			if (max_sourcep->isSyncSlave())
+			{
+				// A sync slave, it doesn't start playing until it's synced up with the master.
+				// Flag this channel as waiting for sync, and return true.
+				channelp->setWaiting(true);
+			}
+			else
+			{
+				channelp->setWaiting(false);
+				channelp->play();
+			}
+		}
+	}
+
+	
+	// Do this BEFORE we update the channels
+	// Update the channels to sync up with any changes that the source made,
+	// such as changing what sound was playing.
+	updateChannels();
+
+	// Update queued sounds (switch to next queued data if the current has finished playing)
+	for (iter = mAllSources.begin(); iter != mAllSources.end(); ++iter)
+	{
+		// This is lame, instead of this I could actually iterate through all the sources
+		// attached to each channel, since only those with active channels
+		// can have anything interesting happen with their queue? (Maybe not true)
+		LLAudioSource *sourcep = iter->second;
+		if (!sourcep->mQueuedDatap)
+		{
+			// Nothing queued, so we don't care.
+			continue;
+		}
+
+		LLAudioChannel *channelp = sourcep->getChannel();
+		if (!channelp)
+		{
+			// This sound isn't playing, so we just process move the queue
+			sourcep->mCurrentDatap = sourcep->mQueuedDatap;
+			sourcep->mQueuedDatap = NULL;
+
+			// Reset the timer so the source doesn't die.
+			sourcep->mAgeTimer.reset();
+			// Make sure we have the buffer set up if we just decoded the data
+			if (sourcep->mCurrentDatap)
+			{
+				updateBufferForData(sourcep->mCurrentDatap);
+			}
+
+			// Actually play the associated data.
+			sourcep->setupChannel();
+			channelp = sourcep->getChannel();
+			if (channelp)
+			{
+				channelp->updateBuffer();
+				sourcep->getChannel()->play();
+			}
+			continue;
+		}
+		else
+		{
+			// Check to see if the current sound is done playing, or looped.
+			if (!channelp->isPlaying())
+			{
+				sourcep->mCurrentDatap = sourcep->mQueuedDatap;
+				sourcep->mQueuedDatap = NULL;
+
+				// Reset the timer so the source doesn't die.
+				sourcep->mAgeTimer.reset();
+
+				// Make sure we have the buffer set up if we just decoded the data
+				if (sourcep->mCurrentDatap)
+				{
+					updateBufferForData(sourcep->mCurrentDatap);
+				}
+
+				// Actually play the associated data.
+				sourcep->setupChannel();
+				channelp->updateBuffer();
+				sourcep->getChannel()->play();
+			}
+			else if (sourcep->isLoop())
+			{
+				// It's a loop, we need to check and see if we're done with it.
+				if (channelp->mLoopedThisFrame)
+				{
+					sourcep->mCurrentDatap = sourcep->mQueuedDatap;
+					sourcep->mQueuedDatap = NULL;
+
+					// Actually, should do a time sync so if we're a loop master/slave
+					// we don't drift away.
+					sourcep->setupChannel();
+					sourcep->getChannel()->play();
+				}
+			}
+		}
+	}
+
+	// Lame, update the channels AGAIN.
+	// Update the channels to sync up with any changes that the source made,
+	// such as changing what sound was playing.
+	updateChannels();
+	
+	// Hack!  For now, just use a global sync master;
+	LLAudioSource *sync_masterp = NULL;
+	LLAudioChannel *master_channelp = NULL;
+	F32 max_sm_priority = -1.f;
+	for (iter = mAllSources.begin(); iter != mAllSources.end(); ++iter)
+	{
+		LLAudioSource *sourcep = iter->second;
+		if (sourcep->isSyncMaster())
+		{
+			if (sourcep->getPriority() > max_sm_priority)
+			{
+				sync_masterp = sourcep;
+				master_channelp = sync_masterp->getChannel();
+				max_sm_priority = sourcep->getPriority();
+			}
+		}
+	}
+
+	if (master_channelp && master_channelp->mLoopedThisFrame)
+	{
+		// Synchronize loop slaves with their masters
+		// Update queued sounds (switch to next queued data if the current has finished playing)
+		for (iter = mAllSources.begin(); iter != mAllSources.end(); ++iter)
+		{
+			LLAudioSource *sourcep = iter->second;
+
+			if (!sourcep->isSyncSlave())
+			{
+				// Not a loop slave, we don't need to do anything
+				continue;
+			}
+
+			LLAudioChannel *channelp = sourcep->getChannel();
+			if (!channelp)
+			{
+				// Not playing, don't need to bother.
+				continue;
+			}
+
+			if (!channelp->isPlaying())
+			{
+				// Now we need to check if our loop master has just looped, and
+				// start playback if that's the case.
+				if (sync_masterp->getChannel())
+				{
+					channelp->playSynced(master_channelp);
+					channelp->setWaiting(false);
+				}
+			}
+		}
+	}
+
+	// Sync up everything that the audio engine needs done.
+	commitDeferredChanges();
+	
+	// Flush unused buffers that are stale enough
+	for (i = 0; i < MAX_BUFFERS; i++)
+	{
+		if (mBuffers[i])
+		{
+			if (!mBuffers[i]->mInUse && mBuffers[i]->mLastUseTimer.getElapsedTimeF32() > 30.f)
+			{
+				//llinfos << "Flushing unused buffer!" << llendl;
+				mBuffers[i]->mAudioDatap->mBufferp = NULL;
+				delete mBuffers[i];
+				mBuffers[i] = NULL;
+			}
+		}
+	}
+
+
+	// Clear all of the looped flags for the channels
+	for (i = 0; i < MAX_CHANNELS; i++)
+	{
+		if (mChannels[i])
+		{
+			mChannels[i]->mLoopedThisFrame = false;
+		}
+	}
+
+	// Decode audio files
+	gAudioDecodeMgrp->processQueue(max_decode_time);
+	
+	// Call this every frame, just in case we somehow
+	// missed picking it up in all the places that can add
+	// or request new data.
+	startNextTransfer();
+
+	updateInternetStream();
+}
+
+
+
+bool LLAudioEngine::updateBufferForData(LLAudioData *adp, const LLUUID &audio_uuid)
+{
+	if (!adp)
+	{
+		return false;
+	}
+
+	// Update the audio buffer first - load a sound if we have it.
+	// Note that this could potentially cause us to waste time updating buffers
+	// for sounds that actually aren't playing, although this should be mitigated
+	// by the fact that we limit the number of buffers, and we flush buffers based
+	// on priority.
+	if (!adp->getBuffer())
+	{
+		if (adp->hasDecodedData())
+		{
+			adp->load();
+		}
+		else if (adp->hasLocalData())
+		{
+			if (audio_uuid.notNull())
+			{
+				gAudioDecodeMgrp->addDecodeRequest(audio_uuid);
+			}
+		}
+		else
+		{
+			return false;
+		}
+	}
+	return true;
+}
+
+
+void LLAudioEngine::enableWind(bool enable)
+{
+	if (enable && (!mEnableWind))
+	{
+		initWind();
+		mEnableWind = enable;
+	}
+	else if (mEnableWind && (!enable))
+	{
+		mEnableWind = enable;
+		cleanupWind();
+	}
+}
+
+
+LLAudioBuffer * LLAudioEngine::getFreeBuffer()
+{
+	S32 i;
+	for (i = 0; i < MAX_BUFFERS; i++)
+	{
+		if (!mBuffers[i])
+		{
+			mBuffers[i] = createBuffer();
+			return mBuffers[i];
+		}
+	}
+
+
+	// Grab the oldest unused buffer
+	F32 max_age = -1.f;
+	S32 buffer_id = -1;
+	for (i = 0; i < MAX_BUFFERS; i++)
+	{
+		if (mBuffers[i])
+		{
+			if (!mBuffers[i]->mInUse)
+			{
+				if (mBuffers[i]->mLastUseTimer.getElapsedTimeF32() > max_age)
+				{
+					max_age = mBuffers[i]->mLastUseTimer.getElapsedTimeF32();
+					buffer_id = i;
+				}
+			}
+		}
+	}
+
+	if (buffer_id >= 0)
+	{
+		llinfos << "Taking over unused buffer " << buffer_id << llendl;
+		//llinfos << "Flushing unused buffer!" << llendl;
+		mBuffers[buffer_id]->mAudioDatap->mBufferp = NULL;
+		delete mBuffers[buffer_id];
+		mBuffers[buffer_id] = createBuffer();
+		return mBuffers[buffer_id];
+	}
+	return NULL;
+}
+
+
+LLAudioChannel * LLAudioEngine::getFreeChannel(const F32 priority)
+{
+	S32 i;
+	for (i = 0; i < mNumChannels; i++)
+	{
+		if (!mChannels[i])
+		{
+			// No channel allocated here, use it.
+			mChannels[i] = createChannel();
+			return mChannels[i];
+		}
+		else
+		{
+			// Channel is allocated but not playing right now, use it.
+			if (!mChannels[i]->isPlaying() && !mChannels[i]->isWaiting())
+			{
+				mChannels[i]->cleanup();
+				if (mChannels[i]->getSource())
+				{
+					mChannels[i]->getSource()->setChannel(NULL);
+				}
+				return mChannels[i];
+			}
+		}
+	}
+
+	// All channels used, check priorities.
+	// Find channel with lowest priority and see if we want to replace it.
+	F32 min_priority = 10000.f;
+	LLAudioChannel *min_channelp = NULL;
+
+	for (i = 0; i < mNumChannels; i++)
+	{
+		LLAudioChannel *channelp = mChannels[i];
+		LLAudioSource *sourcep = channelp->getSource();
+		if (sourcep->getPriority() < min_priority)
+		{
+			min_channelp = channelp;
+			min_priority = sourcep->getPriority();
+		}
+	}
+
+	if (min_priority > priority || !min_channelp)
+	{
+		// All playing channels have higher priority, return.
+		return NULL;
+	}
+
+	// Flush the minimum priority channel, and return it.
+	min_channelp->cleanup();
+	min_channelp->getSource()->setChannel(NULL);
+	return min_channelp;
+}
+
+
+void LLAudioEngine::cleanupBuffer(LLAudioBuffer *bufferp)
+{
+	S32 i;
+	for (i = 0; i < MAX_BUFFERS; i++)
+	{
+		if (mBuffers[i] == bufferp)
+		{
+			delete mBuffers[i];
+			mBuffers[i] = NULL;
+		}
+	}
+}
+
+
+bool LLAudioEngine::preloadSound(const LLUUID &uuid)
+{
+	gAudiop->getAudioData(uuid);	// We don't care about the return value, this is just to make sure
+									// that we have an entry, which will mean that the audio engine knows about this
+
+	if (gAudioDecodeMgrp->addDecodeRequest(uuid))
+	{
+		// This means that we do have a local copy, and we're working on decoding it.
+		return true;
+	}
+
+	// At some point we need to have the audio/asset system check the static VFS
+	// before it goes off and fetches stuff from the server.
+	//llwarns << "Used internal preload for non-local sound" << llendl;
+	return false;
+}
+
+
+bool LLAudioEngine::isWindEnabled()
+{
+	return mEnableWind;
+}
+
+
+void LLAudioEngine::setMuted(bool muted)
+{
+	mMuted = muted;
+	enableWind(!mMuted);
+}
+
+
+void LLAudioEngine::setMasterGain(const F32 gain)
+{
+	mMasterGain = gain;
+	setInternalGain(gain);
+}
+
+F32 LLAudioEngine::getMasterGain()
+{
+	return mMasterGain;
+}
+
+void LLAudioEngine::setSecondaryGain(S32 type, F32 gain)
+{
+	llassert(type < LLAudioEngine::AUDIO_TYPE_COUNT);
+	
+	mSecondaryGain[type] = gain;
+}
+
+F32 LLAudioEngine::getSecondaryGain(S32 type)
+{
+	return mSecondaryGain[type];
+}
+
+F32 LLAudioEngine::getInternetStreamGain()
+{
+	if (mStreamingAudioImpl)
+		return mStreamingAudioImpl->getGain();
+	else
+		return 1.0f;
+}
+
+void LLAudioEngine::setMaxWindGain(F32 gain)
+{
+	mMaxWindGain = gain;
+}
+
+
+F64 LLAudioEngine::mapWindVecToGain(LLVector3 wind_vec)
+{
+	F64 gain = 0.0;
+	
+	gain = wind_vec.magVec();
+
+	if (gain)
+	{
+		if (gain > 20)
+		{
+			gain = 20;
+		}
+		gain = gain/20.0;
+	}
+
+	return (gain);
+} 
+
+
+F64 LLAudioEngine::mapWindVecToPitch(LLVector3 wind_vec)
+{
+	LLVector3 listen_right;
+	F64 theta;
+  
+	// Wind frame is in listener-relative coordinates
+	LLVector3 norm_wind = wind_vec;
+	norm_wind.normVec();
+	listen_right.setVec(1.0,0.0,0.0);
+
+	// measure angle between wind vec and listener right axis (on 0,PI)
+	theta = acos(norm_wind * listen_right);
+
+	// put it on 0, 1
+	theta /= F_PI;					
+
+	// put it on [0, 0.5, 0]
+	if (theta > 0.5) theta = 1.0-theta;
+	if (theta < 0) theta = 0;
+
+	return (theta);
+}
+
+
+F64 LLAudioEngine::mapWindVecToPan(LLVector3 wind_vec)
+{
+	LLVector3 listen_right;
+	F64 theta;
+  
+	// Wind frame is in listener-relative coordinates
+	listen_right.setVec(1.0,0.0,0.0);
+
+	LLVector3 norm_wind = wind_vec;
+	norm_wind.normVec();
+
+	// measure angle between wind vec and listener right axis (on 0,PI)
+	theta = acos(norm_wind * listen_right);
+
+	// put it on 0, 1
+	theta /= F_PI;					
+
+	return (theta);
+}
+
+
+void LLAudioEngine::triggerSound(const LLUUID &audio_uuid, const LLUUID& owner_id, const F32 gain,
+								 const S32 type, const LLVector3d &pos_global)
+{
+	// Create a new source (since this can't be associated with an existing source.
+	//llinfos << "Localized: " << audio_uuid << llendl;
+
+	if (mMuted)
+	{
+		return;
+	}
+
+	LLUUID source_id;
+	source_id.generate();
+
+	LLAudioSource *asp = new LLAudioSource(source_id, owner_id, gain, type);
+	gAudiop->addAudioSource(asp);
+	if (pos_global.isExactlyZero())
+	{
+		asp->setAmbient(true);
+	}
+	else
+	{
+		asp->setPositionGlobal(pos_global);
+	}
+	asp->updatePriority();
+	asp->play(audio_uuid);
+}
+
+
+void LLAudioEngine::setListenerPos(LLVector3 aVec)
+{
+	mListenerp->setPosition(aVec);  
+}
+
+
+LLVector3 LLAudioEngine::getListenerPos()
+{
+	if (mListenerp)
+	{
+		return(mListenerp->getPosition());  
+	}
+	else
+	{
+		return(LLVector3::zero);
+	}
+}
+
+
+void LLAudioEngine::setListenerVelocity(LLVector3 aVec)
+{
+	mListenerp->setVelocity(aVec);  
+}
+
+
+void LLAudioEngine::translateListener(LLVector3 aVec)
+{
+	mListenerp->translate(aVec);	
+}
+
+
+void LLAudioEngine::orientListener(LLVector3 up, LLVector3 at)
+{
+	mListenerp->orient(up, at);  
+}
+
+
+void LLAudioEngine::setListener(LLVector3 pos, LLVector3 vel, LLVector3 up, LLVector3 at)
+{
+	mListenerp->set(pos,vel,up,at);  
+}
+
+
+void LLAudioEngine::setDopplerFactor(F32 factor)
+{
+	if (mListenerp)
+	{
+		mListenerp->setDopplerFactor(factor);  
+	}
+}
+
+
+F32 LLAudioEngine::getDopplerFactor()
+{
+	if (mListenerp)
+	{
+		return mListenerp->getDopplerFactor();
+	}
+	else
+	{
+		return 0.f;
+	}
+}
+
+
+void LLAudioEngine::setRolloffFactor(F32 factor)
+{
+	if (mListenerp)
+	{
+		mListenerp->setRolloffFactor(factor);  
+	}
+}
+
+
+F32 LLAudioEngine::getRolloffFactor()
+{
+	if (mListenerp)
+	{
+		return mListenerp->getRolloffFactor();  
+	}
+	else
+	{
+		return 0.f;
+	}
+}
+
+
+void LLAudioEngine::commitDeferredChanges()
+{
+	mListenerp->commitDeferredChanges();  
+}
+
+
+LLAudioSource * LLAudioEngine::findAudioSource(const LLUUID &source_id)
+{
+	source_map::iterator iter;
+	iter = mAllSources.find(source_id);
+
+	if (iter == mAllSources.end())
+	{
+		return NULL;
+	}
+	else
+	{
+		return iter->second;
+	}
+}
+
+
+LLAudioData * LLAudioEngine::getAudioData(const LLUUID &audio_uuid)
+{
+	data_map::iterator iter;
+	iter = mAllData.find(audio_uuid);
+	if (iter == mAllData.end())
+	{
+		// Create the new audio data
+		LLAudioData *adp = new LLAudioData(audio_uuid);
+		mAllData[audio_uuid] = adp;
+		return adp;
+	}
+	else
+	{
+		return iter->second;
+	}
+}
+
+void LLAudioEngine::addAudioSource(LLAudioSource *asp)
+{
+	mAllSources[asp->getID()] = asp;
+}
+
+
+void LLAudioEngine::cleanupAudioSource(LLAudioSource *asp)
+{
+	source_map::iterator iter;
+	iter = mAllSources.find(asp->getID());
+	if (iter == mAllSources.end())
+	{
+		llwarns << "Cleaning up unknown audio source!" << llendl;
+		return;
+	}
+	delete asp;
+	mAllSources.erase(iter);
+}
+
+
+bool LLAudioEngine::hasDecodedFile(const LLUUID &uuid)
+{
+	std::string uuid_str;
+	uuid.toString(uuid_str);
+
+	std::string wav_path;
+	wav_path = gDirUtilp->getExpandedFilename(LL_PATH_CACHE,uuid_str);
+	wav_path += ".dsf";
+
+	if (gDirUtilp->fileExists(wav_path))
+	{
+		return true;
+	}
+	else
+	{
+		return false;
+	}
+}
+
+
+bool LLAudioEngine::hasLocalFile(const LLUUID &uuid)
+{
+	// See if it's in the VFS.
+	return gVFS->getExists(uuid, LLAssetType::AT_SOUND);
+}
+
+
+void LLAudioEngine::startNextTransfer()
+{
+	//llinfos << "LLAudioEngine::startNextTransfer()" << llendl;
+	if (mCurrentTransfer.notNull() || getMuted())
+	{
+		//llinfos << "Transfer in progress, aborting" << llendl;
+		return;
+	}
+
+	// Get the ID for the next asset that we want to transfer.
+	// Pick one in the following order:
+	LLUUID asset_id;
+	S32 i;
+	LLAudioSource *asp = NULL;
+	LLAudioData *adp = NULL;
+	data_map::iterator data_iter;
+
+	// Check all channels for currently playing sounds.
+	F32 max_pri = -1.f;
+	for (i = 0; i < MAX_CHANNELS; i++)
+	{
+		if (!mChannels[i])
+		{
+			continue;
+		}
+
+		asp = mChannels[i]->getSource();
+		if (!asp)
+		{
+			continue;
+		}
+		if (asp->getPriority() <= max_pri)
+		{
+			continue;
+		}
+
+		if (asp->getPriority() <= max_pri)
+		{
+			continue;
+		}
+
+		adp = asp->getCurrentData();
+		if (!adp)
+		{
+			continue;
+		}
+
+		if (!adp->hasLocalData() && adp->hasValidData())
+		{
+			asset_id = adp->getID();
+			max_pri = asp->getPriority();
+		}
+	}
+
+	// Check all channels for currently queued sounds.
+	if (asset_id.isNull())
+	{
+		max_pri = -1.f;
+		for (i = 0; i < MAX_CHANNELS; i++)
+		{
+			if (!mChannels[i])
+			{
+				continue;
+			}
+
+			LLAudioSource *asp;
+			asp = mChannels[i]->getSource();
+			if (!asp)
+			{
+				continue;
+			}
+
+			if (asp->getPriority() <= max_pri)
+			{
+				continue;
+			}
+
+			adp = asp->getQueuedData();
+			if (!adp)
+			{
+				continue;
+			}
+
+			if (!adp->hasLocalData() && adp->hasValidData())
+			{
+				asset_id = adp->getID();
+				max_pri = asp->getPriority();
+			}
+		}
+	}
+
+	// Check all live channels for other sounds (preloads).
+	if (asset_id.isNull())
+	{
+		max_pri = -1.f;
+		for (i = 0; i < MAX_CHANNELS; i++)
+		{
+			if (!mChannels[i])
+			{
+				continue;
+			}
+
+			LLAudioSource *asp;
+			asp = mChannels[i]->getSource();
+			if (!asp)
+			{
+				continue;
+			}
+
+			if (asp->getPriority() <= max_pri)
+			{
+				continue;
+			}
+
+
+			for (data_iter = asp->mPreloadMap.begin(); data_iter != asp->mPreloadMap.end(); data_iter++)
+			{
+				LLAudioData *adp = data_iter->second;
+				if (!adp)
+				{
+					continue;
+				}
+
+				if (!adp->hasLocalData() && adp->hasValidData())
+				{
+					asset_id = adp->getID();
+					max_pri = asp->getPriority();
+				}
+			}
+		}
+	}
+
+	// Check all sources
+	if (asset_id.isNull())
+	{
+		max_pri = -1.f;
+		source_map::iterator source_iter;
+		for (source_iter = mAllSources.begin(); source_iter != mAllSources.end(); source_iter++)
+		{
+			asp = source_iter->second;
+			if (!asp)
+			{
+				continue;
+			}
+
+			if (asp->getPriority() <= max_pri)
+			{
+				continue;
+			}
+
+			adp = asp->getCurrentData();
+			if (adp && !adp->hasLocalData() && adp->hasValidData())
+			{
+				asset_id = adp->getID();
+				max_pri = asp->getPriority();
+				continue;
+			}
+
+			adp = asp->getQueuedData();
+			if (adp && !adp->hasLocalData() && adp->hasValidData())
+			{
+				asset_id = adp->getID();
+				max_pri = asp->getPriority();
+				continue;
+			}
+
+			for (data_iter = asp->mPreloadMap.begin(); data_iter != asp->mPreloadMap.end(); data_iter++)
+			{
+				LLAudioData *adp = data_iter->second;
+				if (!adp)
+				{
+					continue;
+				}
+
+				if (!adp->hasLocalData() && adp->hasValidData())
+				{
+					asset_id = adp->getID();
+					max_pri = asp->getPriority();
+					break;
+				}
+			}
+		}
+	}
+
+	if (asset_id.notNull())
+	{
+		llinfos << "Getting asset data for: " << asset_id << llendl;
+		gAudiop->mCurrentTransfer = asset_id;
+		gAudiop->mCurrentTransferTimer.reset();
+		gAssetStorage->getAssetData(asset_id, LLAssetType::AT_SOUND,
+									assetCallback, NULL);
+	}
+	else
+	{
+		//llinfos << "No pending transfers?" << llendl;
+	}
+}
+
+
+// static
+void LLAudioEngine::assetCallback(LLVFS *vfs, const LLUUID &uuid, LLAssetType::EType type, void *user_data, S32 result_code, LLExtStat ext_status)
+{
+	if (result_code)
+	{
+		llinfos << "Boom, error in audio file transfer: " << LLAssetStorage::getErrorString( result_code ) << " (" << result_code << ")" << llendl;
+		// Need to mark data as bad to avoid constant rerequests.
+		LLAudioData *adp = gAudiop->getAudioData(uuid);
+		if (adp)
+        {
+			adp->setHasValidData(false);
+			adp->setHasLocalData(false);
+			adp->setHasDecodedData(false);
+		}
+	}
+	else
+	{
+		LLAudioData *adp = gAudiop->getAudioData(uuid);
+		if (!adp)
+        {
+			// Should never happen
+			llwarns << "Got asset callback without audio data for " << uuid << llendl;
+        }
+		else
+		{
+			adp->setHasValidData(true);
+		    adp->setHasLocalData(true);
+		    gAudioDecodeMgrp->addDecodeRequest(uuid);
+		}
+	}
+	gAudiop->mCurrentTransfer = LLUUID::null;
+	gAudiop->startNextTransfer();
+}
+
+
+//
+// LLAudioSource implementation
+//
+
+
+LLAudioSource::LLAudioSource(const LLUUID& id, const LLUUID& owner_id, const F32 gain, const S32 type)
+:	mID(id),
+	mOwnerID(owner_id),
+	mPriority(0.f),
+	mGain(gain),
+	mType(type),
+	mAmbient(false),
+	mLoop(false),
+	mSyncMaster(false),
+	mSyncSlave(false),
+	mQueueSounds(false),
+	mPlayedOnce(false),
+	mChannelp(NULL),
+	mCurrentDatap(NULL),
+	mQueuedDatap(NULL)
+{
+}
+
+
+LLAudioSource::~LLAudioSource()
+{
+	if (mChannelp)
+	{
+		// Stop playback of this sound
+		mChannelp->setSource(NULL);
+		mChannelp = NULL;
+	}
+}
+
+
+void LLAudioSource::setChannel(LLAudioChannel *channelp)
+{
+	if (channelp == mChannelp)
+	{
+		return;
+	}
+
+	mChannelp = channelp;
+}
+
+
+void LLAudioSource::update()
+{
+	if (!getCurrentBuffer())
+	{
+		if (getCurrentData())
+		{
+			// Hack - try and load the sound.  Will do this as a callback
+			// on decode later.
+			if (getCurrentData()->load())
+			{
+				play(getCurrentData()->getID());
+			}			
+		}
+	}
+}
+
+void LLAudioSource::updatePriority()
+{
+	if (isAmbient())
+	{
+		mPriority = 1.f;
+	}
+	else
+	{
+		// Priority is based on distance
+		LLVector3 dist_vec;
+		dist_vec.setVec(getPositionGlobal());
+		dist_vec -= gAudiop->getListenerPos();
+		F32 dist_squared = llmax(1.f, dist_vec.magVecSquared());
+
+		mPriority = mGain / dist_squared;
+	}
+}
+
+bool LLAudioSource::setupChannel()
+{
+	LLAudioData *adp = getCurrentData();
+
+	if (!adp->getBuffer())
+	{
+		// We're not ready to play back the sound yet, so don't try and allocate a channel for it.
+		//llwarns << "Aborting, no buffer" << llendl;
+		return false;
+	}
+
+
+	if (!mChannelp)
+	{
+		// Update the priority, in case we need to push out another channel.
+		updatePriority();
+
+		setChannel(gAudiop->getFreeChannel(getPriority()));
+	}
+
+	if (!mChannelp)
+	{
+		// Ugh, we don't have any free channels.
+		// Now we have to reprioritize.
+		// For now, just don't play the sound.
+		//llwarns << "Aborting, no free channels" << llendl;
+		return false;
+	}
+
+	mChannelp->setSource(this);
+	return true;
+}
+
+
+bool LLAudioSource::play(const LLUUID &audio_uuid)
+{
+	if (audio_uuid.isNull())
+	{
+		if (getChannel())
+		{
+			getChannel()->setSource(NULL);
+			setChannel(NULL);
+			addAudioData(NULL, true);
+		}
+	}
+	// Reset our age timeout if someone attempts to play the source.
+	mAgeTimer.reset();
+
+	LLAudioData *adp = gAudiop->getAudioData(audio_uuid);
+
+	bool has_buffer = gAudiop->updateBufferForData(adp, audio_uuid);
+
+
+	addAudioData(adp);
+
+	if (!has_buffer)
+	{
+		// Don't bother trying to set up a channel or anything, we don't have an audio buffer.
+		return false;
+	}
+
+	if (!setupChannel())
+	{
+		return false;
+	}
+
+	if (isSyncSlave())
+	{
+		// A sync slave, it doesn't start playing until it's synced up with the master.
+		// Flag this channel as waiting for sync, and return true.
+		getChannel()->setWaiting(true);
+		return true;
+	}
+
+	getChannel()->play();
+	return true;
+}
+
+
+bool LLAudioSource::isDone()
+{
+	const F32 MAX_AGE = 60.f;
+	const F32 MAX_UNPLAYED_AGE = 15.f;
+
+	if (isLoop())
+	{
+		// Looped sources never die on their own.
+		return false;
+	}
+
+
+	if (hasPendingPreloads())
+	{
+		return false;
+	}
+
+	if (mQueuedDatap)
+	{
+		// Don't kill this sound if we've got something queued up to play.
+		return false;
+	}
+
+	F32 elapsed = mAgeTimer.getElapsedTimeF32();
+
+	// This is a single-play source
+	if (!mChannelp)
+	{
+		if ((elapsed > MAX_UNPLAYED_AGE) || mPlayedOnce)
+		{
+			// We don't have a channel assigned, and it's been
+			// over 5 seconds since we tried to play it.  Don't bother.
+			//llinfos << "No channel assigned, source is done" << llendl;
+			return true;
+		}
+		else
+		{
+			return false;
+		}
+	}
+
+	if (mChannelp->isPlaying())
+	{
+		if (elapsed > MAX_AGE)
+		{
+			// Arbitarily cut off non-looped sounds when they're old.
+			return true;
+		}
+		else
+		{
+			// Sound is still playing and we haven't timed out, don't kill it.
+			return false;
+		}
+	}
+
+	if ((elapsed > MAX_UNPLAYED_AGE) || mPlayedOnce)
+	{
+		// The sound isn't playing back after 5 seconds or we're already done playing it, kill it.
+		return true;
+	}
+
+	return false;
+}
+
+
+void LLAudioSource::addAudioData(LLAudioData *adp, const bool set_current)
+{
+	// Only handle a single piece of audio data associated with a source right now,
+	// until I implement prefetch.
+	if (set_current)
+	{
+		if (!mCurrentDatap)
+		{
+			mCurrentDatap = adp;
+			if (mChannelp)
+			{
+				mChannelp->updateBuffer();
+				mChannelp->play();
+			}
+
+			// Make sure the audio engine knows that we want to request this sound.
+			gAudiop->startNextTransfer();
+			return;
+		}
+		else if (mQueueSounds)
+		{
+			// If we have current data, and we're queuing, put
+			// the object onto the queue.
+			if (mQueuedDatap)
+			{
+				// We only queue one sound at a time, and it's a FIFO.
+				// Don't put it onto the queue.
+				return;
+			}
+
+			if (adp == mCurrentDatap && isLoop())
+			{
+				// No point in queueing the same sound if
+				// we're looping.
+				return;
+			}
+			mQueuedDatap = adp;
+
+			// Make sure the audio engine knows that we want to request this sound.
+			gAudiop->startNextTransfer();
+		}
+		else
+		{
+			if (mCurrentDatap != adp)
+			{
+				// Right now, if we're currently playing this sound in a channel, we
+				// update the buffer that the channel's associated with
+				// and play it.  This may not be the correct behavior.
+				mCurrentDatap = adp;
+				if (mChannelp)
+				{
+					mChannelp->updateBuffer();
+					mChannelp->play();
+				}
+				// Make sure the audio engine knows that we want to request this sound.
+				gAudiop->startNextTransfer();
+			}
+		}
+	}
+	else
+	{
+		// Add it to the preload list.
+		mPreloadMap[adp->getID()] = adp;
+		gAudiop->startNextTransfer();
+	}
+}
+
+
+bool LLAudioSource::hasPendingPreloads() const
+{
+	// Check to see if we've got any preloads on deck for this source
+	data_map::const_iterator iter;
+	for (iter = mPreloadMap.begin(); iter != mPreloadMap.end(); iter++)
+	{
+		LLAudioData *adp = iter->second;
+		// note: a bad UUID will forever be !hasDecodedData()
+		// but also !hasValidData(), hence the check for hasValidData()
+		if (!adp->hasDecodedData() && adp->hasValidData())
+		{
+			// This source is still waiting for a preload
+			return true;
+		}
+	}
+
+	return false;
+}
+
+
+LLAudioData * LLAudioSource::getCurrentData()
+{
+	return mCurrentDatap;
+}
+
+LLAudioData * LLAudioSource::getQueuedData()
+{
+	return mQueuedDatap;
+}
+
+LLAudioBuffer * LLAudioSource::getCurrentBuffer()
+{
+	if (!mCurrentDatap)
+	{
+		return NULL;
+	}
+
+	return mCurrentDatap->getBuffer();
+}
+
+
+
+
+//
+// LLAudioChannel implementation
+//
+
+
+LLAudioChannel::LLAudioChannel() :
+	mCurrentSourcep(NULL),
+	mCurrentBufferp(NULL),
+	mLoopedThisFrame(false),
+	mWaiting(false),
+	mSecondaryGain(1.0f)
+{
+}
+
+
+LLAudioChannel::~LLAudioChannel()
+{
+	// Need to disconnect any sources which are using this channel.
+	//llinfos << "Cleaning up audio channel" << llendl;
+	if (mCurrentSourcep)
+	{
+		mCurrentSourcep->setChannel(NULL);
+	}
+	mCurrentBufferp = NULL;
+}
+
+
+void LLAudioChannel::setSource(LLAudioSource *sourcep)
+{
+	//llinfos << this << ": setSource(" << sourcep << ")" << llendl;
+
+	if (!sourcep)
+	{
+		// Clearing the source for this channel, don't need to do anything.
+		//llinfos << "Clearing source for channel" << llendl;
+		cleanup();
+		mCurrentSourcep = NULL;
+		mWaiting = false;
+		return;
+	}
+
+	if (sourcep == mCurrentSourcep)
+	{
+		// Don't reallocate the channel, this will make FMOD goofy.
+		//llinfos << "Calling setSource with same source!" << llendl;
+	}
+
+	mCurrentSourcep = sourcep;
+
+
+	updateBuffer();
+	update3DPosition();
+}
+
+
+bool LLAudioChannel::updateBuffer()
+{
+	if (!mCurrentSourcep)
+	{
+		// This channel isn't associated with any source, nothing
+		// to be updated
+		return false;
+	}
+
+	// Initialize the channel's gain setting for this sound.
+	if(gAudiop)
+	{
+		setSecondaryGain(gAudiop->getSecondaryGain(mCurrentSourcep->getType()));
+	}
+
+	LLAudioBuffer *bufferp = mCurrentSourcep->getCurrentBuffer();
+	if (bufferp == mCurrentBufferp)
+	{
+		if (bufferp)
+		{
+			// The source hasn't changed what buffer it's playing
+			bufferp->mLastUseTimer.reset();
+			bufferp->mInUse = true;
+		}
+		return false;
+	}
+
+	//
+	// The source changed what buffer it's playing.  We need to clean up
+	// the existing channel
+	//
+	cleanup();
+
+	mCurrentBufferp = bufferp;
+	if (bufferp)
+	{
+		bufferp->mLastUseTimer.reset();
+		bufferp->mInUse = true;
+	}
+
+	if (!mCurrentBufferp)
+	{
+		// There's no new buffer to be played, so we just abort.
+		return false;
+	}
+
+	return true;
+}
+
+
+
+
+//
+// LLAudioData implementation
+//
+
+
+LLAudioData::LLAudioData(const LLUUID &uuid) :
+	mID(uuid),
+	mBufferp(NULL),
+	mHasLocalData(false),
+	mHasDecodedData(false),
+	mHasValidData(true)
+{
+	if (uuid.isNull())
+	{
+		// This is a null sound.
+		return;
+	}
+	
+	if (gAudiop && gAudiop->hasDecodedFile(uuid))
+	{
+		// Already have a decoded version, don't need to decode it.
+		mHasLocalData = true;
+		mHasDecodedData = true;
+	}
+	else if (gAssetStorage && gAssetStorage->hasLocalAsset(uuid, LLAssetType::AT_SOUND))
+	{
+		mHasLocalData = true;
+	}
+}
+
+
+bool LLAudioData::load()
+{
+	// For now, just assume we're going to use one buffer per audiodata.
+	if (mBufferp)
+	{
+		// We already have this sound in a buffer, don't do anything.
+		llinfos << "Already have a buffer for this sound, don't bother loading!" << llendl;
+		return true;
+	}
+	
+	mBufferp = gAudiop->getFreeBuffer();
+	if (!mBufferp)
+	{
+		// No free buffers, abort.
+		llinfos << "Not able to allocate a new audio buffer, aborting." << llendl;
+		return false;
+	}
+
+	std::string uuid_str;
+	std::string wav_path;
+	mID.toString(uuid_str);
+	wav_path= gDirUtilp->getExpandedFilename(LL_PATH_CACHE,uuid_str) + ".dsf";
+
+	if (!mBufferp->loadWAV(wav_path))
+	{
+		// Hrm.  Right now, let's unset the buffer, since it's empty.
+		gAudiop->cleanupBuffer(mBufferp);
+		mBufferp = NULL;
+
+		return false;
+	}
+	mBufferp->mAudioDatap = this;
+	return true;
+}
+
+
diff --git a/indra/llaudio/llaudioengine.h b/indra/llaudio/llaudioengine.h
new file mode 100644
index 0000000000..457fd93abe
--- /dev/null
+++ b/indra/llaudio/llaudioengine.h
@@ -0,0 +1,452 @@
+/** 
+ * @file audioengine.h
+ * @brief Definition of LLAudioEngine base class abstracting the audio support
+ *
+ * $LicenseInfo:firstyear=2000&license=viewergpl$
+ * 
+ * Copyright (c) 2000-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+
+#ifndef LL_AUDIOENGINE_H
+#define LL_AUDIOENGINE_H
+
+#include <list>
+#include <map>
+
+#include "v3math.h"
+#include "v3dmath.h"
+#include "lltimer.h"
+#include "lluuid.h"
+#include "llframetimer.h"
+#include "llassettype.h"
+
+#include "lllistener.h"
+
+const F32 LL_WIND_UPDATE_INTERVAL = 0.1f;
+const F32 LL_ROLLOFF_MULTIPLIER_UNDER_WATER = 5.f;			//  How much sounds are weaker under water
+const F32 LL_WIND_UNDERWATER_CENTER_FREQ = 20.f;
+
+const F32 ATTACHED_OBJECT_TIMEOUT = 5.0f;
+const F32 DEFAULT_MIN_DISTANCE = 2.0f;
+
+#define MAX_CHANNELS 30
+#define MAX_BUFFERS 40	// Some extra for preloading, maybe?
+
+// This define is intended to allow us to switch from os based wav
+// file loading to vfs based wav file loading. The problem is that I
+// am unconvinced that the LLWaveFile works for loading sounds from
+// memory. So, until that is fixed up, changed, whatever, this remains
+// undefined.
+//#define USE_WAV_VFILE
+
+class LLVFS;
+
+class LLAudioSource;
+class LLAudioData;
+class LLAudioChannel;
+class LLAudioChannelOpenAL;
+class LLAudioBuffer;
+class LLStreamingAudioInterface;
+
+
+//
+//  LLAudioEngine definition
+//
+
+class LLAudioEngine 
+{
+	friend class LLAudioChannelOpenAL; // bleh. channel needs some listener methods.
+	
+public:
+	enum LLAudioType
+	{
+		AUDIO_TYPE_NONE    = 0,
+		AUDIO_TYPE_SFX     = 1,
+		AUDIO_TYPE_UI      = 2,
+		AUDIO_TYPE_AMBIENT = 3,
+		AUDIO_TYPE_COUNT   = 4 // last
+	};
+	
+	LLAudioEngine();
+	virtual ~LLAudioEngine();
+
+	// initialization/startup/shutdown
+	virtual bool init(const S32 num_channels, void *userdata);
+	virtual std::string getDriverName(bool verbose) = 0;
+	virtual void shutdown();
+
+	// Used by the mechanics of the engine
+	//virtual void processQueue(const LLUUID &sound_guid);
+	virtual void setListener(LLVector3 pos,LLVector3 vel,LLVector3 up,LLVector3 at);
+	virtual void updateWind(LLVector3 direction, F32 camera_height_above_water) = 0;
+	virtual void idle(F32 max_decode_time = 0.f);
+	virtual void updateChannels();
+
+	//
+	// "End user" functionality
+	//
+	virtual bool isWindEnabled();
+	virtual void enableWind(bool state_b);
+
+	// Use these for temporarily muting the audio system.
+	// Does not change buffers, initialization, etc. but
+	// stops playing new sounds.
+	virtual void setMuted(bool muted);
+	virtual bool getMuted() const { return mMuted; }
+#ifdef USE_PLUGIN_MEDIA
+	LLPluginClassMedia* initializeMedia(const std::string& media_type);
+#endif
+	F32 getMasterGain();
+	void setMasterGain(F32 gain);
+
+	F32 getSecondaryGain(S32 type);
+	void setSecondaryGain(S32 type, F32 gain);
+
+	F32 getInternetStreamGain();
+
+	virtual void setDopplerFactor(F32 factor);
+	virtual F32 getDopplerFactor();
+	virtual void setRolloffFactor(F32 factor);
+	virtual F32 getRolloffFactor();
+	virtual void setMaxWindGain(F32 gain);
+
+
+	// Methods actually related to setting up and removing sounds
+	// Owner ID is the owner of the object making the request
+	void triggerSound(const LLUUID &sound_id, const LLUUID& owner_id, const F32 gain,
+					  const S32 type = LLAudioEngine::AUDIO_TYPE_NONE,
+					  const LLVector3d &pos_global = LLVector3d::zero);
+	bool preloadSound(const LLUUID &id);
+
+	void addAudioSource(LLAudioSource *asp);
+	void cleanupAudioSource(LLAudioSource *asp);
+
+	LLAudioSource *findAudioSource(const LLUUID &source_id);
+	LLAudioData *getAudioData(const LLUUID &audio_uuid);
+
+	// Internet stream implementation manipulation
+	LLStreamingAudioInterface *getStreamingAudioImpl();
+	void setStreamingAudioImpl(LLStreamingAudioInterface *impl);
+	// Internet stream methods - these will call down into the *mStreamingAudioImpl if it exists
+	void startInternetStream(const std::string& url);
+	void stopInternetStream();
+	void pauseInternetStream(int pause);
+	void updateInternetStream(); // expected to be called often
+	int isInternetStreamPlaying();
+	// use a value from 0.0 to 1.0, inclusive
+	void setInternetStreamGain(F32 vol);
+	std::string getInternetStreamURL();
+	
+	// For debugging usage
+	virtual LLVector3 getListenerPos();
+
+	LLAudioBuffer *getFreeBuffer(); // Get a free buffer, or flush an existing one if you have to.
+	LLAudioChannel *getFreeChannel(const F32 priority); // Get a free channel or flush an existing one if your priority is higher
+	void cleanupBuffer(LLAudioBuffer *bufferp);
+
+	bool hasDecodedFile(const LLUUID &uuid);
+	bool hasLocalFile(const LLUUID &uuid);
+
+	bool updateBufferForData(LLAudioData *adp, const LLUUID &audio_uuid = LLUUID::null);
+
+
+	// Asset callback when we're retrieved a sound from the asset server.
+	void startNextTransfer();
+	static void assetCallback(LLVFS *vfs, const LLUUID &uuid, LLAssetType::EType type, void *user_data, S32 result_code, LLExtStat ext_status);
+
+	friend class LLPipeline; // For debugging
+public:
+	F32 mMaxWindGain; // Hack.  Public to set before fade in?
+
+protected:
+	virtual LLAudioBuffer *createBuffer() = 0;
+	virtual LLAudioChannel *createChannel() = 0;
+
+	virtual void initWind() = 0;
+	virtual void cleanupWind() = 0;
+	virtual void setInternalGain(F32 gain) = 0;
+
+	void commitDeferredChanges();
+
+	virtual void allocateListener() = 0;
+
+
+	// listener methods
+	virtual void setListenerPos(LLVector3 vec);
+	virtual void setListenerVelocity(LLVector3 vec);
+	virtual void orientListener(LLVector3 up, LLVector3 at);
+	virtual void translateListener(LLVector3 vec);
+
+
+	F64 mapWindVecToGain(LLVector3 wind_vec);
+	F64 mapWindVecToPitch(LLVector3 wind_vec);
+	F64 mapWindVecToPan(LLVector3 wind_vec);
+
+protected:
+	LLListener *mListenerp;
+
+	bool mMuted;
+	void* mUserData;
+
+	S32 mLastStatus;
+	
+	S32 mNumChannels;
+	bool mEnableWind;
+
+	LLUUID mCurrentTransfer; // Audio file currently being transferred by the system
+	LLFrameTimer mCurrentTransferTimer;
+
+	// A list of all audio sources that are known to the viewer at this time.
+	// This is most likely a superset of the ones that we actually have audio
+	// data for, or are playing back.
+	typedef std::map<LLUUID, LLAudioSource *> source_map;
+	typedef std::map<LLUUID, LLAudioData *> data_map;
+
+	source_map mAllSources;
+	data_map mAllData;
+
+	LLAudioChannel *mChannels[MAX_CHANNELS];
+
+	// Buffers needs to change into a different data structure, as the number of buffers
+	// that we have active should be limited by RAM usage, not count.
+	LLAudioBuffer *mBuffers[MAX_BUFFERS];
+	
+	F32 mMasterGain;
+	F32 mSecondaryGain[AUDIO_TYPE_COUNT];
+
+	F32 mNextWindUpdate;
+
+	LLFrameTimer mWindUpdateTimer;
+
+private:
+	void setDefaults();
+	LLStreamingAudioInterface *mStreamingAudioImpl;
+};
+
+
+
+
+//
+// Standard audio source.  Can be derived from for special sources, such as those attached to objects.
+//
+
+
+class LLAudioSource
+{
+public:
+	// owner_id is the id of the agent responsible for making this sound
+	// play, for example, the owner of the object currently playing it
+	LLAudioSource(const LLUUID &id, const LLUUID& owner_id, const F32 gain, const S32 type = LLAudioEngine::AUDIO_TYPE_NONE);
+	virtual ~LLAudioSource();
+
+	virtual void update();						// Update this audio source
+	void updatePriority();
+
+	void preload(const LLUUID &audio_id); // Only used for preloading UI sounds, now.
+
+	void addAudioData(LLAudioData *adp, bool set_current = TRUE);
+
+	void setAmbient(const bool ambient)						{ mAmbient = ambient; }
+	bool isAmbient() const									{ return mAmbient; }
+
+	void setLoop(const bool loop)							{ mLoop = loop; }
+	bool isLoop() const										{ return mLoop; }
+
+	void setSyncMaster(const bool master)					{ mSyncMaster = master; }
+	bool isSyncMaster() const								{ return mSyncMaster; }
+
+	void setSyncSlave(const bool slave)						{ mSyncSlave = slave; }
+	bool isSyncSlave() const								{ return mSyncSlave; }
+
+	void setQueueSounds(const bool queue)					{ mQueueSounds = queue; }
+	bool isQueueSounds() const								{ return mQueueSounds; }
+
+	void setPlayedOnce(const bool played_once)				{ mPlayedOnce = played_once; }
+
+	void setType(S32 type)                                  { mType = type; }
+	S32 getType()                                           { return mType; }
+
+	void setPositionGlobal(const LLVector3d &position_global)		{ mPositionGlobal = position_global; }
+	LLVector3d getPositionGlobal() const							{ return mPositionGlobal; }
+	LLVector3 getVelocity()	const									{ return mVelocity; }				
+	F32 getPriority() const											{ return mPriority; }
+
+	// Gain should always be clamped between 0 and 1.
+	F32 getGain() const												{ return mGain; }
+	virtual void setGain(const F32 gain)							{ mGain = llclamp(gain, 0.f, 1.f); }
+
+	const LLUUID &getID() const		{ return mID; }
+	bool isDone();
+
+	LLAudioData *getCurrentData();
+	LLAudioData *getQueuedData();
+	LLAudioBuffer *getCurrentBuffer();
+
+	bool setupChannel();
+	bool play(const LLUUID &audio_id);	// Start the audio source playing
+
+	bool hasPendingPreloads() const;	// Has preloads that haven't been done yet
+
+	friend class LLAudioEngine;
+	friend class LLAudioChannel;
+protected:
+	void setChannel(LLAudioChannel *channelp);
+	LLAudioChannel *getChannel() const						{ return mChannelp; }
+
+protected:
+	LLUUID			mID; // The ID of the source is that of the object if it's attached to an object.
+	LLUUID			mOwnerID;	// owner of the object playing the sound
+	F32				mPriority;
+	F32				mGain;
+	bool			mAmbient;
+	bool			mLoop;
+	bool			mSyncMaster;
+	bool			mSyncSlave;
+	bool			mQueueSounds;
+	bool			mPlayedOnce;
+	S32             mType;
+	LLVector3d		mPositionGlobal;
+	LLVector3		mVelocity;
+
+	//LLAudioSource	*mSyncMasterp;	// If we're a slave, the source that we're synced to.
+	LLAudioChannel	*mChannelp;		// If we're currently playing back, this is the channel that we're assigned to.
+	LLAudioData		*mCurrentDatap;
+	LLAudioData		*mQueuedDatap;
+
+	typedef std::map<LLUUID, LLAudioData *> data_map;
+	data_map mPreloadMap;
+
+	LLFrameTimer mAgeTimer;
+};
+
+
+
+
+//
+// Generic metadata about a particular piece of audio data.
+// The actual data is handled by the derived LLAudioBuffer classes which are
+// derived for each audio engine.
+//
+
+
+class LLAudioData
+{
+public:
+	LLAudioData(const LLUUID &uuid);
+	bool load();
+
+	LLUUID getID() const				{ return mID; }
+	LLAudioBuffer *getBuffer() const	{ return mBufferp; }
+
+	bool	hasLocalData() const		{ return mHasLocalData; }
+	bool	hasDecodedData() const		{ return mHasDecodedData; }
+	bool	hasValidData() const		{ return mHasValidData; }
+
+	void	setHasLocalData(const bool hld)		{ mHasLocalData = hld; }
+	void	setHasDecodedData(const bool hdd)	{ mHasDecodedData = hdd; }
+	void	setHasValidData(const bool hvd)		{ mHasValidData = hvd; }
+
+	friend class LLAudioEngine; // Severe laziness, bad.
+
+protected:
+	LLUUID mID;
+	LLAudioBuffer *mBufferp;	// If this data is being used by the audio system, a pointer to the buffer will be set here.
+	bool mHasLocalData;
+	bool mHasDecodedData;
+	bool mHasValidData;
+};
+
+
+//
+// Base class for an audio channel, i.e. a channel which is capable of playing back a sound.
+// Management of channels is done generically, methods for actually manipulating the channel
+// are derived for each audio engine.
+//
+
+
+class LLAudioChannel
+{
+public:
+	LLAudioChannel();
+	virtual ~LLAudioChannel();
+
+	virtual void setSource(LLAudioSource *sourcep);
+	LLAudioSource *getSource() const			{ return mCurrentSourcep; }
+
+	void setSecondaryGain(F32 gain)             { mSecondaryGain = gain; }
+	F32 getSecondaryGain()                      { return mSecondaryGain; }
+
+	friend class LLAudioEngine;
+	friend class LLAudioSource;
+protected:
+	virtual void play() = 0;
+	virtual void playSynced(LLAudioChannel *channelp) = 0;
+	virtual void cleanup() = 0;
+	virtual bool isPlaying() = 0;
+	void setWaiting(const bool waiting)			{ mWaiting = waiting; }
+	bool isWaiting() const						{ return mWaiting; }
+
+	virtual bool updateBuffer(); // Check to see if the buffer associated with the source changed, and update if necessary.
+	virtual void update3DPosition() = 0;
+	virtual void updateLoop() = 0; // Update your loop/completion status, for use by queueing/syncing.
+protected:
+	LLAudioSource	*mCurrentSourcep;
+	LLAudioBuffer	*mCurrentBufferp;
+	bool			mLoopedThisFrame;
+	bool			mWaiting;	// Waiting for sync.
+	F32             mSecondaryGain;
+};
+
+
+
+
+// Basically an interface class to the engine-specific implementation
+// of audio data that's ready for playback.
+// Will likely get more complex as we decide to do stuff like real streaming audio.
+
+
+class LLAudioBuffer
+{
+public:
+	virtual ~LLAudioBuffer() {};
+	virtual bool loadWAV(const std::string& filename) = 0;
+	virtual U32 getLength() = 0;
+
+	friend class LLAudioEngine;
+	friend class LLAudioChannel;
+	friend class LLAudioData;
+protected:
+	bool mInUse;
+	LLAudioData *mAudioDatap;
+	LLFrameTimer mLastUseTimer;
+};
+
+
+
+extern LLAudioEngine* gAudiop;
+
+#endif
diff --git a/indra/llaudio/llaudioengine_fmod.cpp b/indra/llaudio/llaudioengine_fmod.cpp
new file mode 100644
index 0000000000..7b12b62d53
--- /dev/null
+++ b/indra/llaudio/llaudioengine_fmod.cpp
@@ -0,0 +1,766 @@
+/** 
+ * @file audioengine_fmod.cpp
+ * @brief Implementation of LLAudioEngine class abstracting the audio support as a FMOD 3D implementation
+ *
+ * $LicenseInfo:firstyear=2002&license=viewergpl$
+ * 
+ * Copyright (c) 2002-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#include "linden_common.h"
+
+#include "llstreamingaudio.h"
+#include "llstreamingaudio_fmod.h"
+
+#include "llaudioengine_fmod.h"
+#include "lllistener_fmod.h"
+
+#include "llerror.h"
+#include "llmath.h"
+#include "llrand.h"
+
+#include "fmod.h"
+#include "fmod_errors.h"
+#include "lldir.h"
+#include "llapr.h"
+
+#include "sound_ids.h"
+
+
+extern "C" {
+	void * F_CALLBACKAPI windCallback(void *originalbuffer, void *newbuffer, int length, void* userdata);
+}
+
+FSOUND_DSPUNIT *gWindDSP = NULL;
+
+
+LLAudioEngine_FMOD::LLAudioEngine_FMOD()
+{
+	mInited = false;
+	mWindGen = NULL;
+}
+
+
+LLAudioEngine_FMOD::~LLAudioEngine_FMOD()
+{
+}
+
+
+bool LLAudioEngine_FMOD::init(const S32 num_channels, void* userdata)
+{
+	LLAudioEngine::init(num_channels, userdata);
+
+	// Reserve one extra channel for the http stream.
+	if (!FSOUND_SetMinHardwareChannels(num_channels + 1))
+	{
+		LL_WARNS("AppInit") << "FMOD::init[0](), error: " << FMOD_ErrorString(FSOUND_GetError()) << LL_ENDL;
+	}
+
+	LL_DEBUGS("AppInit") << "LLAudioEngine_FMOD::init() initializing FMOD" << LL_ENDL;
+
+	F32 version = FSOUND_GetVersion();
+	if (version < FMOD_VERSION)
+	{
+		LL_WARNS("AppInit") << "Error : You are using the wrong FMOD version (" << version
+			<< ")!  You should be using FMOD " << FMOD_VERSION << LL_ENDL;
+		//return false;
+	}
+
+	U32 fmod_flags = 0x0;
+
+#if LL_WINDOWS
+	// Windows needs to know which window is frontmost.
+	// This must be called before FSOUND_Init() per the FMOD docs.
+	// This could be used to let FMOD handle muting when we lose focus,
+	// but we don't actually want to do that because we want to distinguish
+	// between minimized and not-focused states.
+	if (!FSOUND_SetHWND(userdata))
+	{
+		LL_WARNS("AppInit") << "Error setting FMOD window: "
+			<< FMOD_ErrorString(FSOUND_GetError()) << LL_ENDL;
+		return false;
+	}
+	// Play audio when we don't have focus.
+	// (For example, IM client on top of us.)
+	// This means we also try to play audio when minimized,
+	// so we manually handle muting in that case. JC
+	fmod_flags |= FSOUND_INIT_GLOBALFOCUS;
+#endif
+
+#if LL_LINUX
+	// initialize the FMOD engine
+
+	// This is a hack to use only FMOD's basic FPU mixer
+	// when the LL_VALGRIND environmental variable is set,
+	// otherwise valgrind will fall over on FMOD's MMX detection
+	if (getenv("LL_VALGRIND"))		/*Flawfinder: ignore*/
+	{
+		LL_INFOS("AppInit") << "Pacifying valgrind in FMOD init." << LL_ENDL;
+		FSOUND_SetMixer(FSOUND_MIXER_QUALITY_FPU);
+	}
+
+	// If we don't set an output method, Linux FMOD always
+	// decides on OSS and fails otherwise.  So we'll manually
+	// try ESD, then OSS, then ALSA.
+	// Why this order?  See SL-13250, but in short, OSS emulated
+	// on top of ALSA is ironically more reliable than raw ALSA.
+	// Ack, and ESD has more reliable failure modes - but has worse
+	// latency - than all of them, so wins for now.
+	bool audio_ok = false;
+
+	if (!audio_ok)
+		if (NULL == getenv("LL_BAD_FMOD_ESD")) /*Flawfinder: ignore*/
+		{
+			LL_DEBUGS("AppInit") << "Trying ESD audio output..." << LL_ENDL;
+			if(FSOUND_SetOutput(FSOUND_OUTPUT_ESD) &&
+			   FSOUND_Init(44100, num_channels, fmod_flags))
+			{
+				LL_DEBUGS("AppInit") << "ESD audio output initialized OKAY"
+					<< LL_ENDL;
+				audio_ok = true;
+			} else {
+				LL_WARNS("AppInit") << "ESD audio output FAILED to initialize: "
+					<< FMOD_ErrorString(FSOUND_GetError()) << LL_ENDL;
+			}
+		} else {
+			LL_DEBUGS("AppInit") << "ESD audio output SKIPPED" << LL_ENDL;
+		}
+
+	if (!audio_ok)
+		if (NULL == getenv("LL_BAD_FMOD_OSS")) 	 /*Flawfinder: ignore*/
+		{
+			LL_DEBUGS("AppInit") << "Trying OSS audio output..."	<< LL_ENDL;
+			if(FSOUND_SetOutput(FSOUND_OUTPUT_OSS) &&
+			   FSOUND_Init(44100, num_channels, fmod_flags))
+			{
+				LL_DEBUGS("AppInit") << "OSS audio output initialized OKAY" << LL_ENDL;
+				audio_ok = true;
+			} else {
+				LL_WARNS("AppInit") << "OSS audio output FAILED to initialize: "
+					<< FMOD_ErrorString(FSOUND_GetError()) << LL_ENDL;
+			}
+		} else {
+			LL_DEBUGS("AppInit") << "OSS audio output SKIPPED" << LL_ENDL;
+		}
+
+	if (!audio_ok)
+		if (NULL == getenv("LL_BAD_FMOD_ALSA"))		/*Flawfinder: ignore*/
+		{
+			LL_DEBUGS("AppInit") << "Trying ALSA audio output..." << LL_ENDL;
+			if(FSOUND_SetOutput(FSOUND_OUTPUT_ALSA) &&
+			   FSOUND_Init(44100, num_channels, fmod_flags))
+			{
+				LL_DEBUGS("AppInit") << "ALSA audio output initialized OKAY" << LL_ENDL;
+				audio_ok = true;
+			} else {
+				LL_WARNS("AppInit") << "ALSA audio output FAILED to initialize: "
+					<< FMOD_ErrorString(FSOUND_GetError()) << LL_ENDL;
+			}
+		} else {
+			LL_DEBUGS("AppInit") << "OSS audio output SKIPPED" << LL_ENDL;
+		}
+
+	if (!audio_ok)
+	{
+		LL_WARNS("AppInit") << "Overall audio init failure." << LL_ENDL;
+		return false;
+	}
+
+	// On Linux, FMOD causes a SIGPIPE for some netstream error
+	// conditions (an FMOD bug); ignore SIGPIPE so it doesn't crash us.
+	// NOW FIXED in FMOD 3.x since 2006-10-01.
+	//signal(SIGPIPE, SIG_IGN);
+
+	// We're interested in logging which output method we
+	// ended up with, for QA purposes.
+	switch (FSOUND_GetOutput())
+	{
+	case FSOUND_OUTPUT_NOSOUND: LL_DEBUGS("AppInit") << "Audio output: NoSound" << LL_ENDL; break;
+	case FSOUND_OUTPUT_OSS:	LL_DEBUGS("AppInit") << "Audio output: OSS" << LL_ENDL; break;
+	case FSOUND_OUTPUT_ESD:	LL_DEBUGS("AppInit") << "Audio output: ESD" << LL_ENDL; break;
+	case FSOUND_OUTPUT_ALSA: LL_DEBUGS("AppInit") << "Audio output: ALSA" << LL_ENDL; break;
+	default: LL_INFOS("AppInit") << "Audio output: Unknown!" << LL_ENDL; break;
+	};
+
+#else // LL_LINUX
+
+	// initialize the FMOD engine
+	if (!FSOUND_Init(44100, num_channels, fmod_flags))
+	{
+		LL_WARNS("AppInit") << "Error initializing FMOD: "
+			<< FMOD_ErrorString(FSOUND_GetError()) << LL_ENDL;
+		return false;
+	}
+	
+#endif
+
+	// set up our favourite FMOD-native streaming audio implementation if none has already been added
+	if (!getStreamingAudioImpl()) // no existing implementation added
+		setStreamingAudioImpl(new LLStreamingAudio_FMOD());
+
+	LL_DEBUGS("AppInit") << "LLAudioEngine_FMOD::init() FMOD initialized correctly" << LL_ENDL;
+
+	mInited = true;
+
+	return true;
+}
+
+
+std::string LLAudioEngine_FMOD::getDriverName(bool verbose)
+{
+	if (verbose)
+	{
+		F32 version = FSOUND_GetVersion();
+		return llformat("FMOD version %f", version);
+	}
+	else
+	{
+		return "FMOD";
+	}
+}
+
+
+void LLAudioEngine_FMOD::allocateListener(void)
+{	
+	mListenerp = (LLListener *) new LLListener_FMOD();
+	if (!mListenerp)
+	{
+		llwarns << "Listener creation failed" << llendl;
+	}
+}
+
+
+void LLAudioEngine_FMOD::shutdown()
+{
+	if (gWindDSP)
+	{
+		FSOUND_DSP_SetActive(gWindDSP,false);
+		FSOUND_DSP_Free(gWindDSP);
+	}
+
+	stopInternetStream();
+
+	LLAudioEngine::shutdown();
+	
+	llinfos << "LLAudioEngine_FMOD::shutdown() closing FMOD" << llendl;
+	FSOUND_Close();
+	llinfos << "LLAudioEngine_FMOD::shutdown() done closing FMOD" << llendl;
+
+	delete mListenerp;
+	mListenerp = NULL;
+}
+
+
+LLAudioBuffer * LLAudioEngine_FMOD::createBuffer()
+{
+	return new LLAudioBufferFMOD();
+}
+
+
+LLAudioChannel * LLAudioEngine_FMOD::createChannel()
+{
+	return new LLAudioChannelFMOD();
+}
+
+
+void LLAudioEngine_FMOD::initWind()
+{
+	mWindGen = new LLWindGen<MIXBUFFERFORMAT>;
+
+	if (!gWindDSP)
+	{
+		gWindDSP = FSOUND_DSP_Create(&windCallback, FSOUND_DSP_DEFAULTPRIORITY_CLEARUNIT + 20, mWindGen);
+	}
+	if (gWindDSP)
+	{
+		FSOUND_DSP_SetActive(gWindDSP, true);
+	}
+	mNextWindUpdate = 0.0;
+}
+
+
+void LLAudioEngine_FMOD::cleanupWind()
+{
+	if (gWindDSP)
+	{
+		FSOUND_DSP_SetActive(gWindDSP, false);
+		FSOUND_DSP_Free(gWindDSP);
+		gWindDSP = NULL;
+	}
+
+	delete mWindGen;
+	mWindGen = NULL;
+}
+
+
+//-----------------------------------------------------------------------
+void LLAudioEngine_FMOD::updateWind(LLVector3 wind_vec, F32 camera_height_above_water)
+{
+	LLVector3 wind_pos;
+	F64 pitch;
+	F64 center_freq;
+
+	if (!mEnableWind)
+	{
+		return;
+	}
+	
+	if (mWindUpdateTimer.checkExpirationAndReset(LL_WIND_UPDATE_INTERVAL))
+	{
+		
+		// wind comes in as Linden coordinate (+X = forward, +Y = left, +Z = up)
+		// need to convert this to the conventional orientation DS3D and OpenAL use
+		// where +X = right, +Y = up, +Z = backwards
+
+		wind_vec.setVec(-wind_vec.mV[1], wind_vec.mV[2], -wind_vec.mV[0]);
+
+		// cerr << "Wind update" << endl;
+
+		pitch = 1.0 + mapWindVecToPitch(wind_vec);
+		center_freq = 80.0 * pow(pitch,2.5*(mapWindVecToGain(wind_vec)+1.0));
+		
+		mWindGen->mTargetFreq = (F32)center_freq;
+		mWindGen->mTargetGain = (F32)mapWindVecToGain(wind_vec) * mMaxWindGain;
+		mWindGen->mTargetPanGainR = (F32)mapWindVecToPan(wind_vec);
+  	}
+}
+
+/*
+//-----------------------------------------------------------------------
+void LLAudioEngine_FMOD::setSourceMinDistance(U16 source_num, F64 distance)
+{
+	if (!mInited)
+	{
+		return;
+	}
+	if (mBuffer[source_num])
+	{
+		mMinDistance[source_num] = (F32) distance;
+		if (!FSOUND_Sample_SetMinMaxDistance(mBuffer[source_num],mMinDistance[source_num], mMaxDistance[source_num]))
+		{
+			llwarns << "FMOD::setSourceMinDistance(" << source_num << "), error: " << FMOD_ErrorString(FSOUND_GetError()) << llendl;
+		}
+	}
+}
+
+//-----------------------------------------------------------------------
+void LLAudioEngine_FMOD::setSourceMaxDistance(U16 source_num, F64 distance)
+{
+	if (!mInited)
+	{
+		return;
+	}
+	if (mBuffer[source_num])
+	{
+		mMaxDistance[source_num] = (F32) distance;
+		if (!FSOUND_Sample_SetMinMaxDistance(mBuffer[source_num],mMinDistance[source_num], mMaxDistance[source_num]))
+		{
+			llwarns << "FMOD::setSourceMaxDistance(" << source_num << "), error: " << FMOD_ErrorString(FSOUND_GetError()) << llendl;
+		}
+	}
+}
+
+//-----------------------------------------------------------------------
+void LLAudioEngine_FMOD::get3DParams(S32 source_num, S32 *volume, S32 *freq, S32 *inside, S32 *outside, LLVector3 *orient, S32 *out_volume, F32 *min_dist, F32 *max_dist)
+{
+	*volume = 0;
+	*freq = 0;
+	*inside = 0;
+	*outside = 0;
+	*orient = LLVector3::zero;
+	*out_volume = 0;
+	*min_dist = 0.f;
+	*max_dist = 0.f;
+}
+
+*/
+
+
+//-----------------------------------------------------------------------
+void LLAudioEngine_FMOD::setInternalGain(F32 gain)
+{
+	if (!mInited)
+	{
+		return;
+	}
+
+	gain = llclamp( gain, 0.0f, 1.0f );
+	FSOUND_SetSFXMasterVolume( llround( 255.0f * gain ) );
+
+	LLStreamingAudioInterface *saimpl = getStreamingAudioImpl();
+	if ( saimpl )
+	{
+		// fmod likes its streaming audio channel gain re-asserted after
+		// master volume change.
+		saimpl->setGain(saimpl->getGain());
+	}
+}
+
+//
+// LLAudioChannelFMOD implementation
+//
+
+LLAudioChannelFMOD::LLAudioChannelFMOD() : LLAudioChannel(), mChannelID(0), mLastSamplePos(0)
+{
+}
+
+
+LLAudioChannelFMOD::~LLAudioChannelFMOD()
+{
+	cleanup();
+}
+
+
+bool LLAudioChannelFMOD::updateBuffer()
+{
+	if (LLAudioChannel::updateBuffer())
+	{
+		// Base class update returned true, which means that we need to actually
+		// set up the channel for a different buffer.
+
+		LLAudioBufferFMOD *bufferp = (LLAudioBufferFMOD *)mCurrentSourcep->getCurrentBuffer();
+
+		// Grab the FMOD sample associated with the buffer
+		FSOUND_SAMPLE *samplep = bufferp->getSample();
+		if (!samplep)
+		{
+			// This is bad, there should ALWAYS be a sample associated with a legit
+			// buffer.
+			llerrs << "No FMOD sample!" << llendl;
+			return false;
+		}
+
+
+		// Actually play the sound.  Start it off paused so we can do all the necessary
+		// setup.
+		mChannelID = FSOUND_PlaySoundEx(FSOUND_FREE, samplep, FSOUND_DSP_GetSFXUnit(), true);
+
+		//llinfos << "Setting up channel " << std::hex << mChannelID << std::dec << llendl;
+	}
+
+	// If we have a source for the channel, we need to update its gain.
+	if (mCurrentSourcep)
+	{
+		// SJB: warnings can spam and hurt framerate, disabling
+		if (!FSOUND_SetVolume(mChannelID, llround(getSecondaryGain() * mCurrentSourcep->getGain() * 255.0f)))
+		{
+// 			llwarns << "LLAudioChannelFMOD::updateBuffer error: " << FMOD_ErrorString(FSOUND_GetError()) << llendl;
+		}
+		
+		if (!FSOUND_SetLoopMode(mChannelID, mCurrentSourcep->isLoop() ? FSOUND_LOOP_NORMAL : FSOUND_LOOP_OFF))
+		{
+// 			llwarns << "Channel " << mChannelID << "Source ID: " << mCurrentSourcep->getID()
+// 					<< " at " << mCurrentSourcep->getPositionGlobal() << llendl;
+// 			llwarns << "LLAudioChannelFMOD::updateBuffer error: " << FMOD_ErrorString(FSOUND_GetError()) << llendl;
+		}
+	}
+
+	return true;
+}
+
+
+void LLAudioChannelFMOD::update3DPosition()
+{
+	if (!mChannelID)
+	{
+		// We're not actually a live channel (i.e., we're not playing back anything)
+		return;
+	}
+
+	LLAudioBufferFMOD *bufferp = (LLAudioBufferFMOD *)mCurrentBufferp;
+	if (!bufferp)
+	{
+		// We don't have a buffer associated with us (should really have been picked up
+		// by the above if.
+		return;
+	}
+
+	if (mCurrentSourcep->isAmbient())
+	{
+		// Ambient sound, don't need to do any positional updates.
+		bufferp->set3DMode(false);
+	}
+	else
+	{
+		// Localized sound.  Update the position and velocity of the sound.
+		bufferp->set3DMode(true);
+
+		LLVector3 float_pos;
+		float_pos.setVec(mCurrentSourcep->getPositionGlobal());
+		if (!FSOUND_3D_SetAttributes(mChannelID, float_pos.mV, mCurrentSourcep->getVelocity().mV))
+		{
+			LL_DEBUGS("FMOD") << "LLAudioChannelFMOD::update3DPosition error: " << FMOD_ErrorString(FSOUND_GetError()) << LL_ENDL;
+		}
+	}
+}
+
+
+void LLAudioChannelFMOD::updateLoop()
+{
+	if (!mChannelID)
+	{
+		// May want to clear up the loop/sample counters.
+		return;
+	}
+
+	//
+	// Hack:  We keep track of whether we looped or not by seeing when the
+	// sample position looks like it's going backwards.  Not reliable; may
+	// yield false negatives.
+	//
+	U32 cur_pos = FSOUND_GetCurrentPosition(mChannelID);
+	if (cur_pos < (U32)mLastSamplePos)
+	{
+		mLoopedThisFrame = true;
+	}
+	mLastSamplePos = cur_pos;
+}
+
+
+void LLAudioChannelFMOD::cleanup()
+{
+	if (!mChannelID)
+	{
+		//llinfos << "Aborting cleanup with no channelID." << llendl;
+		return;
+	}
+
+	//llinfos << "Cleaning up channel: " << mChannelID << llendl;
+	if (!FSOUND_StopSound(mChannelID))
+	{
+		LL_DEBUGS("FMOD") << "LLAudioChannelFMOD::cleanup error: " << FMOD_ErrorString(FSOUND_GetError()) << llendl;
+	}
+
+	mCurrentBufferp = NULL;
+	mChannelID = 0;
+}
+
+
+void LLAudioChannelFMOD::play()
+{
+	if (!mChannelID)
+	{
+		llwarns << "Playing without a channelID, aborting" << llendl;
+		return;
+	}
+
+	if (!FSOUND_SetPaused(mChannelID, false))
+	{
+		llwarns << "LLAudioChannelFMOD::play error: " << FMOD_ErrorString(FSOUND_GetError()) << llendl;
+	}
+	getSource()->setPlayedOnce(true);
+}
+
+
+void LLAudioChannelFMOD::playSynced(LLAudioChannel *channelp)
+{
+	LLAudioChannelFMOD *fmod_channelp = (LLAudioChannelFMOD*)channelp;
+	if (!(fmod_channelp->mChannelID && mChannelID))
+	{
+		// Don't have channels allocated to both the master and the slave
+		return;
+	}
+
+	U32 position = FSOUND_GetCurrentPosition(fmod_channelp->mChannelID) % mCurrentBufferp->getLength();
+	// Try to match the position of our sync master
+	if (!FSOUND_SetCurrentPosition(mChannelID, position))
+	{
+		llwarns << "LLAudioChannelFMOD::playSynced unable to set current position" << llendl;
+	}
+
+	// Start us playing
+	play();
+}
+
+
+bool LLAudioChannelFMOD::isPlaying()
+{
+	if (!mChannelID)
+	{
+		return false;
+	}
+
+	return FSOUND_IsPlaying(mChannelID) && (!FSOUND_GetPaused(mChannelID));
+}
+
+
+
+//
+// LLAudioBufferFMOD implementation
+//
+
+
+LLAudioBufferFMOD::LLAudioBufferFMOD()
+{
+	mSamplep = NULL;
+}
+
+
+LLAudioBufferFMOD::~LLAudioBufferFMOD()
+{
+	if (mSamplep)
+	{
+		// Clean up the associated FMOD sample if it exists.
+		FSOUND_Sample_Free(mSamplep);
+		mSamplep = NULL;
+	}
+}
+
+
+bool LLAudioBufferFMOD::loadWAV(const std::string& filename)
+{
+	// Try to open a wav file from disk.  This will eventually go away, as we don't
+	// really want to block doing this.
+	if (filename.empty())
+	{
+		// invalid filename, abort.
+		return false;
+	}
+
+	if (!LLAPRFile::isExist(filename, NULL, LL_APR_RPB))
+	{
+		// File not found, abort.
+		return false;
+	}
+	
+	if (mSamplep)
+	{
+		// If there's already something loaded in this buffer, clean it up.
+		FSOUND_Sample_Free(mSamplep);
+		mSamplep = NULL;
+	}
+
+	// Load up the wav file into an fmod sample
+#if LL_WINDOWS
+	// MikeS. - Loading the sound file manually and then handing it over to FMOD,
+	//	since FMOD uses posix IO internally,
+	// which doesn't work with unicode file paths.
+	LLFILE* sound_file = LLFile::fopen(filename,"rb");	/* Flawfinder: ignore */
+	if (sound_file)
+	{
+		fseek(sound_file,0,SEEK_END);
+		U32	file_length = ftell(sound_file);	//Find the length of the file by seeking to the end and getting the offset
+		size_t	read_count;
+		fseek(sound_file,0,SEEK_SET);	//Seek back to the beginning
+		char*	buffer = new char[file_length];
+		llassert(buffer);
+		read_count = fread((void*)buffer,file_length,1,sound_file);//Load it..
+		if(ferror(sound_file)==0 && (read_count == 1)){//No read error, and we got 1 chunk of our size...
+			unsigned int mode_flags = FSOUND_LOOP_NORMAL | FSOUND_LOADMEMORY;
+									//FSOUND_16BITS | FSOUND_MONO | FSOUND_LOADMEMORY | FSOUND_LOOP_NORMAL;
+			mSamplep = FSOUND_Sample_Load(FSOUND_UNMANAGED, buffer, mode_flags , 0, file_length);
+		}
+		delete[] buffer;
+		fclose(sound_file);
+	}
+#else
+	mSamplep = FSOUND_Sample_Load(FSOUND_UNMANAGED, filename.c_str(), FSOUND_LOOP_NORMAL, 0, 0);
+#endif
+
+	if (!mSamplep)
+	{
+		// We failed to load the file for some reason.
+		llwarns << "Could not load data '" << filename << "': "
+				<< FMOD_ErrorString(FSOUND_GetError()) << llendl;
+
+		//
+		// If we EVER want to load wav files provided by end users, we need
+		// to rethink this!
+		//
+		// file is probably corrupt - remove it.
+		LLFile::remove(filename);
+		return false;
+	}
+
+	// Everything went well, return true
+	return true;
+}
+
+
+U32 LLAudioBufferFMOD::getLength()
+{
+	if (!mSamplep)
+	{
+		return 0;
+	}
+
+	return FSOUND_Sample_GetLength(mSamplep);
+}
+
+
+void LLAudioBufferFMOD::set3DMode(bool use3d)
+{
+	U16 current_mode = FSOUND_Sample_GetMode(mSamplep);
+	
+	if (use3d)
+	{
+		if (!FSOUND_Sample_SetMode(mSamplep, (current_mode & (~FSOUND_2D))))
+		{
+			llwarns << "LLAudioBufferFMOD::set3DMode error: " << FMOD_ErrorString(FSOUND_GetError()) << llendl;
+		}
+	}
+	else
+	{
+		if (!FSOUND_Sample_SetMode(mSamplep, current_mode | FSOUND_2D))
+		{
+			llwarns << "LLAudioBufferFMOD::set3DMode error: " << FMOD_ErrorString(FSOUND_GetError()) << llendl;
+		}
+	}
+}
+
+
+void * F_CALLBACKAPI windCallback(void *originalbuffer, void *newbuffer, int length, void* userdata)
+{
+	// originalbuffer = fmod's original mixbuffer.
+	// newbuffer = the buffer passed from the previous DSP unit.
+	// length = length in samples at this mix time.
+	// param = user parameter passed through in FSOUND_DSP_Create.
+	//
+	// modify the buffer in some fashion
+
+	LLWindGen<LLAudioEngine_FMOD::MIXBUFFERFORMAT> *windgen =
+		(LLWindGen<LLAudioEngine_FMOD::MIXBUFFERFORMAT> *)userdata;
+	U8 stride;
+
+#if LL_DARWIN
+	stride = sizeof(LLAudioEngine_FMOD::MIXBUFFERFORMAT);
+#else
+	int mixertype = FSOUND_GetMixer();
+	if (mixertype == FSOUND_MIXER_BLENDMODE ||
+	    mixertype == FSOUND_MIXER_QUALITY_FPU)
+	{
+		stride = 4;
+	}
+	else
+	{
+		stride = 2;
+	}
+#endif
+
+	newbuffer = windgen->windGenerate((LLAudioEngine_FMOD::MIXBUFFERFORMAT *)newbuffer, length, stride);
+
+	return newbuffer;
+}
diff --git a/indra/llaudio/llaudioengine_fmod.h b/indra/llaudio/llaudioengine_fmod.h
new file mode 100644
index 0000000000..3968657cba
--- /dev/null
+++ b/indra/llaudio/llaudioengine_fmod.h
@@ -0,0 +1,129 @@
+/** 
+ * @file audioengine_fmod.h
+ * @brief Definition of LLAudioEngine class abstracting the audio
+ * support as a FMOD 3D implementation
+ *
+ * $LicenseInfo:firstyear=2002&license=viewergpl$
+ * 
+ * Copyright (c) 2002-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#ifndef LL_AUDIOENGINE_FMOD_H
+#define LL_AUDIOENGINE_FMOD_H
+
+#include "llaudioengine.h"
+#include "lllistener_fmod.h"
+#include "llwindgen.h"
+
+#include "fmod.h"
+
+class LLAudioStreamManagerFMOD;
+
+class LLAudioEngine_FMOD : public LLAudioEngine 
+{
+public:
+	LLAudioEngine_FMOD();
+	virtual ~LLAudioEngine_FMOD();
+
+	// initialization/startup/shutdown
+	virtual bool init(const S32 num_channels, void *user_data);
+       	virtual std::string getDriverName(bool verbose);
+	virtual void allocateListener();
+
+	virtual void shutdown();
+
+	/*virtual*/ void initWind();
+	/*virtual*/ void cleanupWind();
+
+	/*virtual*/void updateWind(LLVector3 direction, F32 camera_height_above_water);
+
+#if LL_DARWIN
+        typedef S32 MIXBUFFERFORMAT;
+#else
+        typedef S16 MIXBUFFERFORMAT;
+#endif
+
+protected:
+	/*virtual*/ LLAudioBuffer *createBuffer(); // Get a free buffer, or flush an existing one if you have to.
+	/*virtual*/ LLAudioChannel *createChannel(); // Create a new audio channel.
+
+	/*virtual*/ void setInternalGain(F32 gain);
+protected:
+	static signed char F_CALLBACKAPI callbackMetaData(char* name, char* value, void* userdata);
+
+	//F32 mMinDistance[MAX_BUFFERS];
+	//F32 mMaxDistance[MAX_BUFFERS];
+
+	bool mInited;
+
+	// On Windows, userdata is the HWND of the application window.
+	void* mUserData;
+
+	LLWindGen<MIXBUFFERFORMAT> *mWindGen;
+};
+
+
+class LLAudioChannelFMOD : public LLAudioChannel
+{
+public:
+	LLAudioChannelFMOD();
+	virtual ~LLAudioChannelFMOD();
+
+protected:
+	/*virtual*/ void play();
+	/*virtual*/ void playSynced(LLAudioChannel *channelp);
+	/*virtual*/ void cleanup();
+	/*virtual*/ bool isPlaying();
+
+	/*virtual*/ bool updateBuffer();
+	/*virtual*/ void update3DPosition();
+	/*virtual*/ void updateLoop();
+
+protected:
+	int mChannelID;
+	S32 mLastSamplePos;
+};
+
+
+class LLAudioBufferFMOD : public LLAudioBuffer
+{
+public:
+	LLAudioBufferFMOD();
+	virtual ~LLAudioBufferFMOD();
+
+	/*virtual*/ bool loadWAV(const std::string& filename);
+	/*virtual*/ U32 getLength();
+	friend class LLAudioChannelFMOD;
+
+	void set3DMode(bool use3d);
+protected:
+	FSOUND_SAMPLE *getSample()	{ return mSamplep; }
+protected:
+	FSOUND_SAMPLE *mSamplep;
+};
+
+
+#endif // LL_AUDIOENGINE_FMOD_H
diff --git a/indra/llaudio/llaudioengine_openal.cpp b/indra/llaudio/llaudioengine_openal.cpp
new file mode 100644
index 0000000000..a5982ccbd6
--- /dev/null
+++ b/indra/llaudio/llaudioengine_openal.cpp
@@ -0,0 +1,546 @@
+/**
+ * @file audioengine_openal.cpp
+ * @brief implementation of audio engine using OpenAL
+ * support as a OpenAL 3D implementation
+ *
+ * $LicenseInfo:firstyear=2002&license=viewergpl$
+ * 
+ * Copyright (c) 2002-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#include "linden_common.h"
+#include "lldir.h"
+
+#include "llaudioengine_openal.h"
+#include "lllistener_openal.h"
+
+
+LLAudioEngine_OpenAL::LLAudioEngine_OpenAL()
+	:
+	mWindGen(NULL),
+	mWindBuf(NULL),
+	mWindBufFreq(0),
+	mWindBufSamples(0),
+	mWindBufBytes(0),
+	mWindSource(AL_NONE),
+	mNumEmptyWindALBuffers(MAX_NUM_WIND_BUFFERS)
+{
+}
+
+// virtual
+LLAudioEngine_OpenAL::~LLAudioEngine_OpenAL()
+{
+}
+
+// virtual
+bool LLAudioEngine_OpenAL::init(const S32 num_channels, void* userdata)
+{
+	mWindGen = NULL;
+	LLAudioEngine::init(num_channels, userdata);
+
+	if(!alutInit(NULL, NULL))
+	{
+		llwarns << "LLAudioEngine_OpenAL::init() ALUT initialization failed: " << alutGetErrorString (alutGetError ()) << llendl;
+		return false;
+	}
+
+	llinfos << "LLAudioEngine_OpenAL::init() OpenAL successfully initialized" << llendl;
+
+	llinfos << "OpenAL version: "
+		<< ll_safe_string(alGetString(AL_VERSION)) << llendl;
+	llinfos << "OpenAL vendor: "
+		<< ll_safe_string(alGetString(AL_VENDOR)) << llendl;
+	llinfos << "OpenAL renderer: "
+		<< ll_safe_string(alGetString(AL_RENDERER)) << llendl;
+
+	ALint major = alutGetMajorVersion ();
+	ALint minor = alutGetMinorVersion ();
+	llinfos << "ALUT version: " << major << "." << minor << llendl;
+
+	ALCdevice *device = alcGetContextsDevice(alcGetCurrentContext());
+
+	alcGetIntegerv(device, ALC_MAJOR_VERSION, 1, &major);
+	alcGetIntegerv(device, ALC_MAJOR_VERSION, 1, &minor);
+	llinfos << "ALC version: " << major << "." << minor << llendl;
+
+	llinfos << "ALC default device: "
+		<< ll_safe_string(alcGetString(device,
+					       ALC_DEFAULT_DEVICE_SPECIFIER))
+		<< llendl;
+
+	return true;
+}
+
+// virtual
+std::string LLAudioEngine_OpenAL::getDriverName(bool verbose)
+{
+	ALCdevice *device = alcGetContextsDevice(alcGetCurrentContext());
+	std::ostringstream version;
+
+	version <<
+		"OpenAL";
+
+	if (verbose)
+	{
+		version <<
+			", version " <<
+			ll_safe_string(alGetString(AL_VERSION)) <<
+			" / " <<
+			ll_safe_string(alGetString(AL_VENDOR)) <<
+			" / " <<
+			ll_safe_string(alGetString(AL_RENDERER));
+		
+		if (device)
+			version <<
+				": " <<
+				ll_safe_string(alcGetString(device,
+				    ALC_DEFAULT_DEVICE_SPECIFIER));
+	}
+
+	return version.str();
+}
+
+// virtual
+void LLAudioEngine_OpenAL::allocateListener()
+{
+	mListenerp = (LLListener *) new LLListener_OpenAL();
+	if(!mListenerp)
+	{
+		llwarns << "LLAudioEngine_OpenAL::allocateListener() Listener creation failed" << llendl;
+	}
+}
+
+// virtual
+void LLAudioEngine_OpenAL::shutdown()
+{
+	llinfos << "About to LLAudioEngine::shutdown()" << llendl;
+	LLAudioEngine::shutdown();
+
+	llinfos << "About to alutExit()" << llendl;
+	if(!alutExit())
+	{
+		llwarns << "Nuts." << llendl;
+		llwarns << "LLAudioEngine_OpenAL::shutdown() ALUT shutdown failed: " << alutGetErrorString (alutGetError ()) << llendl;
+	}
+
+	llinfos << "LLAudioEngine_OpenAL::shutdown() OpenAL successfully shut down" << llendl;
+
+	delete mListenerp;
+	mListenerp = NULL;
+}
+
+LLAudioBuffer *LLAudioEngine_OpenAL::createBuffer()
+{
+	return new LLAudioBufferOpenAL();
+}
+
+LLAudioChannel *LLAudioEngine_OpenAL::createChannel()
+{
+	return new LLAudioChannelOpenAL();
+}
+
+void LLAudioEngine_OpenAL::setInternalGain(F32 gain)
+{
+	//llinfos << "LLAudioEngine_OpenAL::setInternalGain() Gain: " << gain << llendl;
+	alListenerf(AL_GAIN, gain);
+}
+
+LLAudioChannelOpenAL::LLAudioChannelOpenAL()
+	:
+	mALSource(AL_NONE),
+	mLastSamplePos(0)
+{
+	alGenSources(1, &mALSource);
+}
+
+LLAudioChannelOpenAL::~LLAudioChannelOpenAL()
+{
+	cleanup();
+	alDeleteSources(1, &mALSource);
+}
+
+void LLAudioChannelOpenAL::cleanup()
+{
+	alSourceStop(mALSource);
+	mCurrentBufferp = NULL;
+}
+
+void LLAudioChannelOpenAL::play()
+{
+	if (mALSource == AL_NONE)
+	{
+		llwarns << "Playing without a mALSource, aborting" << llendl;
+		return;
+	}
+
+	if(!isPlaying())
+	{
+		alSourcePlay(mALSource);
+		getSource()->setPlayedOnce(true);
+	}
+}
+
+void LLAudioChannelOpenAL::playSynced(LLAudioChannel *channelp)
+{
+	if (channelp)
+	{
+		LLAudioChannelOpenAL *masterchannelp =
+			(LLAudioChannelOpenAL*)channelp;
+		if (mALSource != AL_NONE &&
+		    masterchannelp->mALSource != AL_NONE)
+		{
+			// we have channels allocated to master and slave
+			ALfloat master_offset;
+			alGetSourcef(masterchannelp->mALSource, AL_SEC_OFFSET,
+				     &master_offset);
+
+			llinfos << "Syncing with master at " << master_offset
+				<< "sec" << llendl;
+			// *TODO: detect when this fails, maybe use AL_SAMPLE_
+			alSourcef(mALSource, AL_SEC_OFFSET, master_offset);
+		}
+	}
+	play();
+}
+
+bool LLAudioChannelOpenAL::isPlaying()
+{
+	if (mALSource != AL_NONE)
+	{
+		ALint state;
+		alGetSourcei(mALSource, AL_SOURCE_STATE, &state);
+		if(state == AL_PLAYING)
+		{
+			return true;
+		}
+	}
+		
+	return false;
+}
+
+bool LLAudioChannelOpenAL::updateBuffer()
+{
+	if (LLAudioChannel::updateBuffer())
+	{
+		// Base class update returned true, which means that we need to actually
+		// set up the source for a different buffer.
+		LLAudioBufferOpenAL *bufferp = (LLAudioBufferOpenAL *)mCurrentSourcep->getCurrentBuffer();
+		ALuint buffer = bufferp->getBuffer();
+		alSourcei(mALSource, AL_BUFFER, buffer);
+		mLastSamplePos = 0;
+	}
+
+	if (mCurrentSourcep)
+	{
+		alSourcef(mALSource, AL_GAIN,
+			  mCurrentSourcep->getGain() * getSecondaryGain());
+		alSourcei(mALSource, AL_LOOPING,
+			  mCurrentSourcep->isLoop() ? AL_TRUE : AL_FALSE);
+		alSourcef(mALSource, AL_ROLLOFF_FACTOR,
+			  gAudiop->mListenerp->getRolloffFactor());
+	}
+
+	return true;
+}
+
+
+void LLAudioChannelOpenAL::updateLoop()
+{
+	if (mALSource == AL_NONE)
+	{
+		return;
+	}
+
+	// Hack:  We keep track of whether we looped or not by seeing when the
+	// sample position looks like it's going backwards.  Not reliable; may
+	// yield false negatives.
+	//
+	ALint cur_pos;
+	alGetSourcei(mALSource, AL_SAMPLE_OFFSET, &cur_pos);
+	if (cur_pos < mLastSamplePos)
+	{
+		mLoopedThisFrame = true;
+	}
+	mLastSamplePos = cur_pos;
+}
+
+
+void LLAudioChannelOpenAL::update3DPosition()
+{
+	if(!mCurrentSourcep)
+	{
+		return;
+	}
+	if (mCurrentSourcep->isAmbient())
+	{
+		alSource3f(mALSource, AL_POSITION, 0.0, 0.0, 0.0);
+		alSource3f(mALSource, AL_VELOCITY, 0.0, 0.0, 0.0);
+		alSourcei (mALSource, AL_SOURCE_RELATIVE, AL_TRUE);
+	} else {
+		LLVector3 float_pos;
+		float_pos.setVec(mCurrentSourcep->getPositionGlobal());
+		alSourcefv(mALSource, AL_POSITION, float_pos.mV);
+		alSourcefv(mALSource, AL_VELOCITY, mCurrentSourcep->getVelocity().mV);
+		alSourcei (mALSource, AL_SOURCE_RELATIVE, AL_FALSE);
+	}
+
+	alSourcef(mALSource, AL_GAIN, mCurrentSourcep->getGain() * getSecondaryGain());
+}
+
+LLAudioBufferOpenAL::LLAudioBufferOpenAL()
+{
+	mALBuffer = AL_NONE;
+}
+
+LLAudioBufferOpenAL::~LLAudioBufferOpenAL()
+{
+	cleanup();
+}
+
+void LLAudioBufferOpenAL::cleanup()
+{
+	if(mALBuffer != AL_NONE)
+	{
+		alDeleteBuffers(1, &mALBuffer);
+		mALBuffer = AL_NONE;
+	}
+}
+
+bool LLAudioBufferOpenAL::loadWAV(const std::string& filename)
+{
+	cleanup();
+	mALBuffer = alutCreateBufferFromFile(filename.c_str());
+	if(mALBuffer == AL_NONE)
+	{
+		ALenum error = alutGetError(); 
+		if (gDirUtilp->fileExists(filename))
+		{
+			llwarns <<
+				"LLAudioBufferOpenAL::loadWAV() Error loading "
+				<< filename
+				<< " " << alutGetErrorString(error) << llendl;
+		}
+		else
+		{
+			// It's common for the file to not actually exist.
+			lldebugs <<
+				"LLAudioBufferOpenAL::loadWAV() Error loading "
+				 << filename
+				 << " " << alutGetErrorString(error) << llendl;
+		}
+		return false;
+	}
+
+	return true;
+}
+
+U32 LLAudioBufferOpenAL::getLength()
+{
+	if(mALBuffer == AL_NONE)
+	{
+		return 0;
+	}
+	ALint length;
+	alGetBufferi(mALBuffer, AL_SIZE, &length);
+	return length / 2; // convert size in bytes to size in (16-bit) samples
+}
+
+// ------------
+
+void LLAudioEngine_OpenAL::initWind()
+{
+	ALenum error;
+	llinfos << "LLAudioEngine_OpenAL::initWind() start" << llendl;
+
+	mNumEmptyWindALBuffers = MAX_NUM_WIND_BUFFERS;
+
+	alGetError(); /* clear error */
+	
+	alGenSources(1,&mWindSource);
+	
+	if((error=alGetError()) != AL_NO_ERROR)
+	{
+		llwarns << "LLAudioEngine_OpenAL::initWind() Error creating wind sources: "<<error<<llendl;
+	}
+
+	mWindGen = new LLWindGen<WIND_SAMPLE_T>;
+
+	mWindBufFreq = mWindGen->getInputSamplingRate();
+	mWindBufSamples = llceil(mWindBufFreq * WIND_BUFFER_SIZE_SEC);
+	mWindBufBytes = mWindBufSamples * 2 /*stereo*/ * sizeof(WIND_SAMPLE_T);
+
+	mWindBuf = new WIND_SAMPLE_T [mWindBufSamples * 2 /*stereo*/];
+
+	if(mWindBuf==NULL)
+	{
+		llerrs << "LLAudioEngine_OpenAL::initWind() Error creating wind memory buffer" << llendl;
+		mEnableWind=false;
+	}
+
+	llinfos << "LLAudioEngine_OpenAL::initWind() done" << llendl;
+}
+
+void LLAudioEngine_OpenAL::cleanupWind()
+{
+	llinfos << "LLAudioEngine_OpenAL::cleanupWind()" << llendl;
+
+	if (mWindSource != AL_NONE)
+	{
+		// detach and delete all outstanding buffers on the wind source
+		alSourceStop(mWindSource);
+		ALint processed;
+		alGetSourcei(mWindSource, AL_BUFFERS_PROCESSED, &processed);
+		while (processed--)
+		{
+			ALuint buffer = AL_NONE;
+			alSourceUnqueueBuffers(mWindSource, 1, &buffer);
+			alDeleteBuffers(1, &buffer);
+		}
+
+		// delete the wind source itself
+		alDeleteSources(1, &mWindSource);
+
+		mWindSource = AL_NONE;
+	}
+	
+	delete[] mWindBuf;
+	mWindBuf = NULL;
+
+	delete mWindGen;
+	mWindGen = NULL;
+}
+
+void LLAudioEngine_OpenAL::updateWind(LLVector3 wind_vec, F32 camera_altitude)
+{
+	LLVector3 wind_pos;
+	F64 pitch;
+	F64 center_freq;
+	ALenum error;
+	
+	if (!mEnableWind)
+		return;
+	
+	if(!mWindBuf)
+		return;
+	
+	if (mWindUpdateTimer.checkExpirationAndReset(LL_WIND_UPDATE_INTERVAL))
+	{
+		
+		// wind comes in as Linden coordinate (+X = forward, +Y = left, +Z = up)
+		// need to convert this to the conventional orientation DS3D and OpenAL use
+		// where +X = right, +Y = up, +Z = backwards
+		
+		wind_vec.setVec(-wind_vec.mV[1], wind_vec.mV[2], -wind_vec.mV[0]);
+		
+		pitch = 1.0 + mapWindVecToPitch(wind_vec);
+		center_freq = 80.0 * pow(pitch,2.5*(mapWindVecToGain(wind_vec)+1.0));
+		
+		mWindGen->mTargetFreq = (F32)center_freq;
+		mWindGen->mTargetGain = (F32)mapWindVecToGain(wind_vec) * mMaxWindGain;
+		mWindGen->mTargetPanGainR = (F32)mapWindVecToPan(wind_vec);
+		
+		alSourcei(mWindSource, AL_LOOPING, AL_FALSE);
+		alSource3f(mWindSource, AL_POSITION, 0.0, 0.0, 0.0);
+		alSource3f(mWindSource, AL_VELOCITY, 0.0, 0.0, 0.0);
+		alSourcef(mWindSource, AL_ROLLOFF_FACTOR, 0.0);
+		alSourcei(mWindSource, AL_SOURCE_RELATIVE, AL_TRUE);
+	}
+
+	// ok lets make a wind buffer now
+
+	ALint processed, queued, unprocessed;
+	alGetSourcei(mWindSource, AL_BUFFERS_PROCESSED, &processed);
+	alGetSourcei(mWindSource, AL_BUFFERS_QUEUED, &queued);
+	unprocessed = queued - processed;
+
+	// ensure that there are always at least 3x as many filled buffers
+	// queued as we managed to empty since last time.
+	mNumEmptyWindALBuffers = llmin(mNumEmptyWindALBuffers + processed * 3 - unprocessed, MAX_NUM_WIND_BUFFERS-unprocessed);
+	mNumEmptyWindALBuffers = llmax(mNumEmptyWindALBuffers, 0);
+
+	//llinfos << "mNumEmptyWindALBuffers: " << mNumEmptyWindALBuffers	<<" (" << unprocessed << ":" << processed << ")" << llendl;
+
+	while(processed--) // unqueue old buffers
+	{
+		ALuint buffer;
+		ALenum error;
+		alGetError(); /* clear error */
+		alSourceUnqueueBuffers(mWindSource, 1, &buffer);
+		error = alGetError();
+		if(error != AL_NO_ERROR)
+		{
+			llwarns << "LLAudioEngine_OpenAL::updateWind() error swapping (unqueuing) buffers" << llendl;
+		}
+		else
+		{
+			alDeleteBuffers(1, &buffer);
+		}
+	}
+
+	unprocessed += mNumEmptyWindALBuffers;
+	while (mNumEmptyWindALBuffers > 0) // fill+queue new buffers
+	{
+		ALuint buffer;
+		alGetError(); /* clear error */
+		alGenBuffers(1,&buffer);
+		if((error=alGetError()) != AL_NO_ERROR)
+		{
+			llwarns << "LLAudioEngine_OpenAL::initWind() Error creating wind buffer: " << error << llendl;
+			break;
+		}
+
+		alBufferData(buffer,
+			     AL_FORMAT_STEREO16,
+			     mWindGen->windGenerate(mWindBuf,
+						    mWindBufSamples, 2),
+			     mWindBufBytes,
+			     mWindBufFreq);
+		error = alGetError();
+		if(error != AL_NO_ERROR)
+		{
+			llwarns << "LLAudioEngine_OpenAL::updateWind() error swapping (bufferdata) buffers" << llendl;
+		}
+		
+		alSourceQueueBuffers(mWindSource, 1, &buffer);
+		error = alGetError();
+		if(error != AL_NO_ERROR)
+		{
+			llwarns << "LLAudioEngine_OpenAL::updateWind() error swapping (queuing) buffers" << llendl;
+		}
+
+		--mNumEmptyWindALBuffers;
+	}
+
+	ALint playing;
+	alGetSourcei(mWindSource, AL_SOURCE_STATE, &playing);
+	if(playing != AL_PLAYING)
+	{
+		alSourcePlay(mWindSource);
+
+		lldebugs << "Wind had stopped - probably ran out of buffers - restarting: " << (unprocessed+mNumEmptyWindALBuffers) << " now queued." << llendl;
+	}
+}
+
diff --git a/indra/llaudio/llaudioengine_openal.h b/indra/llaudio/llaudioengine_openal.h
new file mode 100644
index 0000000000..5aca03e195
--- /dev/null
+++ b/indra/llaudio/llaudioengine_openal.h
@@ -0,0 +1,114 @@
+/**
+ * @file audioengine_openal.cpp
+ * @brief implementation of audio engine using OpenAL
+ * support as a OpenAL 3D implementation
+ *
+ *
+ * $LicenseInfo:firstyear=2002&license=viewergpl$
+ * 
+ * Copyright (c) 2002-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+
+#ifndef LL_AUDIOENGINE_OPENAL_H
+#define LL_AUDIOENGINE_OPENAL_H
+
+#include "llaudioengine.h"
+#include "lllistener_openal.h"
+#include "llwindgen.h"
+
+class LLAudioEngine_OpenAL : public LLAudioEngine
+{
+	public:
+		LLAudioEngine_OpenAL();
+		virtual ~LLAudioEngine_OpenAL();
+
+		virtual bool init(const S32 num_channels, void *user_data);
+        	virtual std::string getDriverName(bool verbose);
+		virtual void allocateListener();
+
+		virtual void shutdown();
+
+		void setInternalGain(F32 gain);
+
+		LLAudioBuffer* createBuffer();
+		LLAudioChannel* createChannel();
+
+		/*virtual*/ void initWind();
+		/*virtual*/ void cleanupWind();
+		/*virtual*/ void updateWind(LLVector3 direction, F32 camera_altitude);
+
+	private:
+		void * windDSP(void *newbuffer, int length);
+	        typedef S16 WIND_SAMPLE_T;
+        	LLWindGen<WIND_SAMPLE_T> *mWindGen;
+        	S16 *mWindBuf;
+        	U32 mWindBufFreq;
+        	U32 mWindBufSamples;
+        	U32 mWindBufBytes;
+        	ALuint mWindSource;
+	        int mNumEmptyWindALBuffers;
+
+        	static const int MAX_NUM_WIND_BUFFERS = 80;
+        	static const float WIND_BUFFER_SIZE_SEC = 0.05f; // 1/20th sec
+};
+
+class LLAudioChannelOpenAL : public LLAudioChannel
+{
+	public:
+		LLAudioChannelOpenAL();
+		virtual ~LLAudioChannelOpenAL();
+	protected:
+		/*virtual*/ void play();
+		/*virtual*/ void playSynced(LLAudioChannel *channelp);
+		/*virtual*/ void cleanup();
+		/*virtual*/ bool isPlaying();
+
+		/*virtual*/ bool updateBuffer();
+		/*virtual*/ void update3DPosition();
+		/*virtual*/ void updateLoop();
+
+		ALuint mALSource;
+	        ALint mLastSamplePos;
+};
+
+class LLAudioBufferOpenAL : public LLAudioBuffer{
+	public:
+		LLAudioBufferOpenAL();
+		virtual ~LLAudioBufferOpenAL();
+
+		bool loadWAV(const std::string& filename);
+		U32 getLength();
+
+		friend class LLAudioChannelOpenAL;
+	protected:
+		void cleanup();
+		ALuint getBuffer() {return mALBuffer;}
+
+		ALuint mALBuffer;
+};
+
+#endif
diff --git a/indra/llaudio/lllistener.cpp b/indra/llaudio/lllistener.cpp
new file mode 100644
index 0000000000..846c6bccb5
--- /dev/null
+++ b/indra/llaudio/lllistener.cpp
@@ -0,0 +1,142 @@
+/** 
+ * @file listener.cpp
+ * @brief Implementation of LISTENER class abstracting the audio support
+ *
+ * $LicenseInfo:firstyear=2000&license=viewergpl$
+ * 
+ * Copyright (c) 2000-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#include "linden_common.h"
+
+#include "lllistener.h"
+
+#define DEFAULT_AT  0.0f,0.0f,-1.0f
+#define DEFAULT_UP  0.0f,1.0f,0.0f
+
+//-----------------------------------------------------------------------
+// constructor
+//-----------------------------------------------------------------------
+LLListener::LLListener()
+{
+	init();
+}
+
+//-----------------------------------------------------------------------
+LLListener::~LLListener()
+{
+}
+
+//-----------------------------------------------------------------------
+void LLListener::init(void)
+{
+	mPosition.zeroVec();
+	mListenAt.setVec(DEFAULT_AT);
+	mListenUp.setVec(DEFAULT_UP);
+	mVelocity.zeroVec();
+}
+
+//-----------------------------------------------------------------------
+void LLListener::translate(LLVector3 offset)
+{
+	mPosition += offset;
+}
+
+//-----------------------------------------------------------------------
+void LLListener::setPosition(LLVector3 pos)
+{
+	mPosition = pos;
+}
+
+//-----------------------------------------------------------------------
+LLVector3 LLListener::getPosition(void)
+{
+	return(mPosition);
+}
+
+//-----------------------------------------------------------------------
+LLVector3 LLListener::getAt(void)
+{
+	return(mListenAt);
+}
+
+//-----------------------------------------------------------------------
+LLVector3 LLListener::getUp(void)
+{
+	return(mListenUp);
+}
+
+//-----------------------------------------------------------------------
+void LLListener::setVelocity(LLVector3 vel)
+{
+	mVelocity = vel;
+}
+
+//-----------------------------------------------------------------------
+void LLListener::orient(LLVector3 up, LLVector3 at)
+{
+	mListenUp = up;
+	mListenAt = at;
+}
+
+//-----------------------------------------------------------------------
+void LLListener::set(LLVector3 pos, LLVector3 vel, LLVector3 up, LLVector3 at)
+{
+	mPosition = pos;
+	mVelocity = vel;
+
+	setPosition(pos);
+	setVelocity(vel);
+	orient(up,at);
+}
+
+//-----------------------------------------------------------------------
+void LLListener::setDopplerFactor(F32 factor)
+{
+}
+
+//-----------------------------------------------------------------------
+F32 LLListener::getDopplerFactor()
+{
+	return (1.f);
+}
+
+//-----------------------------------------------------------------------
+void LLListener::setRolloffFactor(F32 factor)
+{
+}
+
+//-----------------------------------------------------------------------
+F32 LLListener::getRolloffFactor()
+{
+	return (1.f);
+}
+
+//-----------------------------------------------------------------------
+void LLListener::commitDeferredChanges()
+{
+}
+
diff --git a/indra/llaudio/lllistener.h b/indra/llaudio/lllistener.h
new file mode 100644
index 0000000000..e94fbe853f
--- /dev/null
+++ b/indra/llaudio/lllistener.h
@@ -0,0 +1,78 @@
+/** 
+ * @file listener.h
+ * @brief Description of LISTENER base class abstracting the audio support.
+ *
+ * $LicenseInfo:firstyear=2000&license=viewergpl$
+ * 
+ * Copyright (c) 2000-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#ifndef LL_LISTENER_H
+#define LL_LISTENER_H
+
+#include "v3math.h"
+
+class LLListener 
+{
+ private:
+ protected:
+	LLVector3 mPosition;
+	LLVector3 mVelocity;
+	LLVector3 mListenAt;
+	LLVector3 mListenUp;
+
+ public:
+
+ private:
+ protected:
+ public:  
+	LLListener();
+	virtual ~LLListener();
+	virtual void init();  
+
+	virtual void set(LLVector3 pos, LLVector3 vel, LLVector3 up, LLVector3 at);
+
+	virtual void setPosition(LLVector3 pos);
+	virtual void setVelocity(LLVector3 vel);
+
+	virtual void orient(LLVector3 up, LLVector3 at);
+	virtual void translate(LLVector3 offset);
+
+	virtual void setDopplerFactor(F32 factor);
+	virtual void setRolloffFactor(F32 factor);
+
+	virtual LLVector3 getPosition();
+	virtual LLVector3 getAt();
+	virtual LLVector3 getUp();
+
+	virtual F32 getDopplerFactor();
+	virtual F32 getRolloffFactor();
+
+	virtual void commitDeferredChanges();
+};
+
+#endif
+
diff --git a/indra/llaudio/lllistener_ds3d.h b/indra/llaudio/lllistener_ds3d.h
new file mode 100644
index 0000000000..1ff9c170c4
--- /dev/null
+++ b/indra/llaudio/lllistener_ds3d.h
@@ -0,0 +1,74 @@
+/** 
+ * @file listener_ds3d.h
+ * @brief Description of LISTENER class abstracting the audio support
+ * as a DirectSound 3D implementation (windows only)
+ *
+ * $LicenseInfo:firstyear=2000&license=viewergpl$
+ * 
+ * Copyright (c) 2000-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#ifndef LL_LISTENER_DS3D_H
+#define LL_LISTENER_DS3D_H
+
+#include "lllistener.h"
+
+#include <dmusici.h>
+#include <dsound.h>
+#include <ks.h>
+
+class LLListener_DS3D : public LLListener
+{
+ private:
+ protected:
+	IDirectSound3DListener8 *m3DListener;
+ public:
+
+ private:
+ protected:
+ public:  
+	LLListener_DS3D();
+	virtual ~LLListener_DS3D();
+	virtual void init();  
+
+	virtual void setDS3DLPtr (IDirectSound3DListener8 *listener_p);
+
+	virtual void translate(LLVector3 offset);
+	virtual void setPosition(LLVector3 pos);
+	virtual void setVelocity(LLVector3 vel);
+	virtual void orient(LLVector3 up, LLVector3 at);
+
+	virtual void setDopplerFactor(F32 factor);
+	virtual F32 getDopplerFactor();
+	virtual void setRolloffFactor(F32 factor);
+	virtual F32 getRolloffFactor();
+
+	virtual void commitDeferredChanges();
+};
+
+#endif
+
+
diff --git a/indra/llaudio/lllistener_fmod.cpp b/indra/llaudio/lllistener_fmod.cpp
new file mode 100644
index 0000000000..57ad461b02
--- /dev/null
+++ b/indra/llaudio/lllistener_fmod.cpp
@@ -0,0 +1,131 @@
+/** 
+ * @file listener_fmod.cpp
+ * @brief implementation of LISTENER class abstracting the audio
+ * support as a FMOD 3D implementation (windows only)
+ *
+ * $LicenseInfo:firstyear=2002&license=viewergpl$
+ * 
+ * Copyright (c) 2002-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#include "linden_common.h"
+#include "llaudioengine.h"
+#include "lllistener_fmod.h"
+#include "fmod.h"
+
+//-----------------------------------------------------------------------
+// constructor
+//-----------------------------------------------------------------------
+LLListener_FMOD::LLListener_FMOD()
+{
+	init();
+}
+
+//-----------------------------------------------------------------------
+LLListener_FMOD::~LLListener_FMOD()
+{
+}
+
+//-----------------------------------------------------------------------
+void LLListener_FMOD::init(void)
+{
+	// do inherited
+	LLListener::init();
+	mDopplerFactor = 1.0f;
+	mRolloffFactor = 1.0f;
+}
+
+//-----------------------------------------------------------------------
+void LLListener_FMOD::translate(LLVector3 offset)
+{
+	LLListener::translate(offset);
+
+	FSOUND_3D_Listener_SetAttributes(mPosition.mV, NULL, mListenAt.mV[0],mListenAt.mV[1],mListenAt.mV[2], mListenUp.mV[0],mListenUp.mV[1],mListenUp.mV[2]);
+}
+
+//-----------------------------------------------------------------------
+void LLListener_FMOD::setPosition(LLVector3 pos)
+{
+	LLListener::setPosition(pos);
+
+	FSOUND_3D_Listener_SetAttributes(pos.mV, NULL, mListenAt.mV[0],mListenAt.mV[1],mListenAt.mV[2], mListenUp.mV[0],mListenUp.mV[1],mListenUp.mV[2]);
+}
+
+//-----------------------------------------------------------------------
+void LLListener_FMOD::setVelocity(LLVector3 vel)
+{
+	LLListener::setVelocity(vel);
+
+	FSOUND_3D_Listener_SetAttributes(NULL, vel.mV, mListenAt.mV[0],mListenAt.mV[1],mListenAt.mV[2], mListenUp.mV[0],mListenUp.mV[1],mListenUp.mV[2]);
+}
+
+//-----------------------------------------------------------------------
+void LLListener_FMOD::orient(LLVector3 up, LLVector3 at)
+{
+	LLListener::orient(up, at);
+
+	// Welcome to the transition between right and left
+	// (coordinate systems, that is)
+	// Leaving the at vector alone results in a L/R reversal
+	// since DX is left-handed and we (LL, OpenGL, OpenAL) are right-handed
+	at = -at;
+
+	FSOUND_3D_Listener_SetAttributes(NULL, NULL, at.mV[0],at.mV[1],at.mV[2], up.mV[0],up.mV[1],up.mV[2]);
+}
+
+//-----------------------------------------------------------------------
+void LLListener_FMOD::commitDeferredChanges()
+{
+	FSOUND_Update();
+}
+
+
+void LLListener_FMOD::setRolloffFactor(F32 factor)
+{
+	mRolloffFactor = factor;
+	FSOUND_3D_SetRolloffFactor(factor);
+}
+
+
+F32 LLListener_FMOD::getRolloffFactor()
+{
+	return mRolloffFactor;
+}
+
+
+void LLListener_FMOD::setDopplerFactor(F32 factor)
+{
+	mDopplerFactor = factor;
+	FSOUND_3D_SetDopplerFactor(factor);
+}
+
+
+F32 LLListener_FMOD::getDopplerFactor()
+{
+	return mDopplerFactor;
+}
+
+
diff --git a/indra/llaudio/lllistener_fmod.h b/indra/llaudio/lllistener_fmod.h
new file mode 100644
index 0000000000..5a48ec8b68
--- /dev/null
+++ b/indra/llaudio/lllistener_fmod.h
@@ -0,0 +1,64 @@
+/** 
+ * @file listener_fmod.h
+ * @brief Description of LISTENER class abstracting the audio support
+ * as an FMOD 3D implementation (windows and Linux)
+ *
+ * $LicenseInfo:firstyear=2002&license=viewergpl$
+ * 
+ * Copyright (c) 2002-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#ifndef LL_LISTENER_FMOD_H
+#define LL_LISTENER_FMOD_H
+
+#include "lllistener.h"
+
+class LLListener_FMOD : public LLListener
+{
+ public:  
+	LLListener_FMOD();
+	virtual ~LLListener_FMOD();
+	virtual void init();  
+
+	virtual void translate(LLVector3 offset);
+	virtual void setPosition(LLVector3 pos);
+	virtual void setVelocity(LLVector3 vel);
+	virtual void orient(LLVector3 up, LLVector3 at);
+	virtual void commitDeferredChanges();
+
+	virtual void setDopplerFactor(F32 factor);
+	virtual F32 getDopplerFactor();
+	virtual void setRolloffFactor(F32 factor);
+	virtual F32 getRolloffFactor();
+
+ protected:
+	 F32 mDopplerFactor;
+	 F32 mRolloffFactor;
+};
+
+#endif
+
+
diff --git a/indra/llaudio/lllistener_openal.cpp b/indra/llaudio/lllistener_openal.cpp
new file mode 100644
index 0000000000..a96ebd5dba
--- /dev/null
+++ b/indra/llaudio/lllistener_openal.cpp
@@ -0,0 +1,116 @@
+/**
+ * @file audioengine_openal.cpp
+ * @brief implementation of audio engine using OpenAL
+ * support as a OpenAL 3D implementation
+ *
+ * $LicenseInfo:firstyear=2002&license=viewergpl$
+ * 
+ * Copyright (c) 2002-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#include "linden_common.h"
+#include "llaudioengine.h"
+
+#include "lllistener_openal.h"
+
+LLListener_OpenAL::LLListener_OpenAL()
+{
+	init();
+}
+
+LLListener_OpenAL::~LLListener_OpenAL()
+{
+}
+
+void LLListener_OpenAL::translate(LLVector3 offset)
+{
+	//llinfos << "LLListener_OpenAL::translate() : " << offset << llendl;
+	LLListener::translate(offset);
+}
+
+void LLListener_OpenAL::setPosition(LLVector3 pos)
+{
+	//llinfos << "LLListener_OpenAL::setPosition() : " << pos << llendl;
+	LLListener::setPosition(pos);
+}
+
+void LLListener_OpenAL::setVelocity(LLVector3 vel)
+{
+	LLListener::setVelocity(vel);
+}
+
+void LLListener_OpenAL::orient(LLVector3 up, LLVector3 at)
+{
+	//llinfos << "LLListener_OpenAL::orient() up: " << up << " at: " << at << llendl;
+	LLListener::orient(up, at);
+}
+
+void LLListener_OpenAL::commitDeferredChanges()
+{
+	ALfloat orientation[6];
+	orientation[0] = mListenAt.mV[0];
+	orientation[1] = mListenAt.mV[1];
+	orientation[2] = mListenAt.mV[2];
+	orientation[3] = mListenUp.mV[0];
+	orientation[4] = mListenUp.mV[1];
+	orientation[5] = mListenUp.mV[2];
+
+	ALfloat velocity[3];
+	velocity[0] = mVelocity.mV[0];
+	velocity[1] = mVelocity.mV[1];
+	velocity[2] = mVelocity.mV[2];
+
+	alListenerfv(AL_ORIENTATION, orientation);
+	alListenerfv(AL_POSITION, mPosition.mV);
+	alListenerfv(AL_VELOCITY, velocity);
+}
+
+void LLListener_OpenAL::setDopplerFactor(F32 factor)
+{
+	//llinfos << "LLListener_OpenAL::setDopplerFactor() : " << factor << llendl;
+	alDopplerFactor(factor);
+}
+
+F32 LLListener_OpenAL::getDopplerFactor()
+{
+	ALfloat factor;
+	factor = alGetFloat(AL_DOPPLER_FACTOR);
+	//llinfos << "LLListener_OpenAL::getDopplerFactor() : " << factor << llendl;
+	return factor;
+}
+
+
+void LLListener_OpenAL::setRolloffFactor(F32 factor)
+{
+	mRolloffFactor = factor;
+}
+
+F32 LLListener_OpenAL::getRolloffFactor()
+{
+	return mRolloffFactor;
+}
+
+
diff --git a/indra/llaudio/lllistener_openal.h b/indra/llaudio/lllistener_openal.h
new file mode 100644
index 0000000000..0dfeea5c90
--- /dev/null
+++ b/indra/llaudio/lllistener_openal.h
@@ -0,0 +1,64 @@
+/** 
+ * @file listener_openal.h
+ * @brief Description of LISTENER class abstracting the audio support
+ * as an OpenAL implementation
+ *
+ * $LicenseInfo:firstyear=2000&license=viewergpl$
+ * 
+ * Copyright (c) 2000-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#ifndef LL_LISTENER_OPENAL_H
+#define LL_LISTENER_OPENAL_H
+
+#include "lllistener.h"
+
+#include "AL/al.h"
+#include "AL/alut.h"
+
+class LLListener_OpenAL  : public LLListener
+{
+ public:  
+	LLListener_OpenAL();
+	virtual ~LLListener_OpenAL();
+
+	virtual void translate(LLVector3 offset);
+	virtual void setPosition(LLVector3 pos);
+	virtual void setVelocity(LLVector3 vel);
+	virtual void orient(LLVector3 up, LLVector3 at);
+	virtual void commitDeferredChanges();
+
+	virtual void setDopplerFactor(F32 factor);
+	virtual F32 getDopplerFactor();
+	virtual void setRolloffFactor(F32 factor);
+	virtual F32 getRolloffFactor();
+
+ protected:
+	F32 mRolloffFactor;
+};
+
+#endif
+
diff --git a/indra/llaudio/llstreamingaudio.h b/indra/llaudio/llstreamingaudio.h
new file mode 100644
index 0000000000..aa89e6a177
--- /dev/null
+++ b/indra/llaudio/llstreamingaudio.h
@@ -0,0 +1,56 @@
+/** 
+ * @file streamingaudio.h
+ * @author Tofu Linden
+ * @brief Definition of LLStreamingAudioInterface base class abstracting the streaming audio interface
+ *
+ * $LicenseInfo:firstyear=2009&license=viewergpl$
+ * 
+ * Copyright (c) 2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#ifndef LL_STREAMINGAUDIO_H
+#define LL_STREAMINGAUDIO_H
+
+#include "stdtypes.h" // from llcommon
+
+// Entirely abstract.  Based exactly on the historic API.
+class LLStreamingAudioInterface
+{
+ public:
+	virtual ~LLStreamingAudioInterface() {}
+
+	virtual void start(const std::string& url) = 0;
+	virtual void stop() = 0;
+	virtual void pause(int pause) = 0;
+	virtual void update() = 0;
+	virtual int isPlaying() = 0;
+	// use a value from 0.0 to 1.0, inclusive
+	virtual void setGain(F32 vol) = 0;
+	virtual F32 getGain() = 0;
+	virtual std::string getURL() = 0;
+};
+
+#endif // LL_STREAMINGAUDIO_H
diff --git a/indra/llaudio/llstreamingaudio_fmod.cpp b/indra/llaudio/llstreamingaudio_fmod.cpp
new file mode 100644
index 0000000000..a71a87203c
--- /dev/null
+++ b/indra/llaudio/llstreamingaudio_fmod.cpp
@@ -0,0 +1,362 @@
+/** 
+ * @file streamingaudio_fmod.cpp
+ * @brief LLStreamingAudio_FMOD implementation
+ *
+ * $LicenseInfo:firstyear=2009&license=viewergpl$
+ * 
+ * Copyright (c) 2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#include "linden_common.h"
+
+#include "llmath.h"
+
+#include "fmod.h"
+#include "fmod_errors.h"
+
+#include "llstreamingaudio_fmod.h"
+
+
+class LLAudioStreamManagerFMOD
+{
+public:
+	LLAudioStreamManagerFMOD(const std::string& url);
+	int	startStream();
+	bool stopStream(); // Returns true if the stream was successfully stopped.
+	bool ready();
+
+	const std::string& getURL() 	{ return mInternetStreamURL; }
+
+	int getOpenState();
+protected:
+	FSOUND_STREAM* mInternetStream;
+	bool mReady;
+
+	std::string mInternetStreamURL;
+};
+
+
+
+//---------------------------------------------------------------------------
+// Internet Streaming
+//---------------------------------------------------------------------------
+LLStreamingAudio_FMOD::LLStreamingAudio_FMOD() :
+	mCurrentInternetStreamp(NULL),
+	mFMODInternetStreamChannel(-1),
+	mGain(1.0f)
+{
+	// Number of milliseconds of audio to buffer for the audio card.
+	// Must be larger than the usual Second Life frame stutter time.
+	FSOUND_Stream_SetBufferSize(200);
+
+	// Here's where we set the size of the network buffer and some buffering 
+	// parameters.  In this case we want a network buffer of 16k, we want it 
+	// to prebuffer 40% of that when we first connect, and we want it 
+	// to rebuffer 80% of that whenever we encounter a buffer underrun.
+
+	// Leave the net buffer properties at the default.
+	//FSOUND_Stream_Net_SetBufferProperties(20000, 40, 80);
+}
+
+
+LLStreamingAudio_FMOD::~LLStreamingAudio_FMOD()
+{
+	// nothing interesting/safe to do.
+}
+
+
+void LLStreamingAudio_FMOD::start(const std::string& url)
+{
+	//if (!mInited)
+	//{
+	//	llwarns << "startInternetStream before audio initialized" << llendl;
+	//	return;
+	//}
+
+	// "stop" stream but don't clear url, etc. in case url == mInternetStreamURL
+	stop();
+
+	if (!url.empty())
+	{
+		llinfos << "Starting internet stream: " << url << llendl;
+		mCurrentInternetStreamp = new LLAudioStreamManagerFMOD(url);
+		mURL = url;
+	}
+	else
+	{
+		llinfos << "Set internet stream to null" << llendl;
+		mURL.clear();
+	}
+}
+
+
+void LLStreamingAudio_FMOD::update()
+{
+	// Kill dead internet streams, if possible
+	std::list<LLAudioStreamManagerFMOD *>::iterator iter;
+	for (iter = mDeadStreams.begin(); iter != mDeadStreams.end();)
+	{
+		LLAudioStreamManagerFMOD *streamp = *iter;
+		if (streamp->stopStream())
+		{
+			llinfos << "Closed dead stream" << llendl;
+			delete streamp;
+			mDeadStreams.erase(iter++);
+		}
+		else
+		{
+			iter++;
+		}
+	}
+
+	// Don't do anything if there are no streams playing
+	if (!mCurrentInternetStreamp)
+	{
+		return;
+	}
+
+	int open_state = mCurrentInternetStreamp->getOpenState();
+
+	if (!open_state)
+	{
+		// Stream is live
+
+		// start the stream if it's ready
+		if (mFMODInternetStreamChannel < 0)
+		{
+			mFMODInternetStreamChannel = mCurrentInternetStreamp->startStream();
+
+			if (mFMODInternetStreamChannel != -1)
+			{
+				// Reset volume to previously set volume
+				setGain(getGain());
+				FSOUND_SetPaused(mFMODInternetStreamChannel, false);
+			}
+		}
+	}
+		
+	switch(open_state)
+	{
+	default:
+	case 0:
+		// success
+		break;
+	case -1:
+		// stream handle is invalid
+		llwarns << "InternetStream - invalid handle" << llendl;
+		stop();
+		return;
+	case -2:
+		// opening
+		break;
+	case -3:
+		// failed to open, file not found, perhaps
+		llwarns << "InternetSteam - failed to open" << llendl;
+		stop();
+		return;
+	case -4:
+		// connecting
+		break;
+	case -5:
+		// buffering
+		break;
+	}
+
+}
+
+void LLStreamingAudio_FMOD::stop()
+{
+	if (mFMODInternetStreamChannel != -1)
+	{
+		FSOUND_SetPaused(mFMODInternetStreamChannel, true);
+		FSOUND_SetPriority(mFMODInternetStreamChannel, 0);
+		mFMODInternetStreamChannel = -1;
+	}
+
+	if (mCurrentInternetStreamp)
+	{
+		llinfos << "Stopping internet stream: " << mCurrentInternetStreamp->getURL() << llendl;
+		if (mCurrentInternetStreamp->stopStream())
+		{
+			delete mCurrentInternetStreamp;
+		}
+		else
+		{
+			llwarns << "Pushing stream to dead list: " << mCurrentInternetStreamp->getURL() << llendl;
+			mDeadStreams.push_back(mCurrentInternetStreamp);
+		}
+		mCurrentInternetStreamp = NULL;
+		//mURL.clear();
+	}
+}
+
+void LLStreamingAudio_FMOD::pause(int pauseopt)
+{
+	if (pauseopt < 0)
+	{
+		pauseopt = mCurrentInternetStreamp ? 1 : 0;
+	}
+
+	if (pauseopt)
+	{
+		if (mCurrentInternetStreamp)
+		{
+			stop();
+		}
+	}
+	else
+	{
+		start(getURL());
+	}
+}
+
+
+// A stream is "playing" if it has been requested to start.  That
+// doesn't necessarily mean audio is coming out of the speakers.
+int LLStreamingAudio_FMOD::isPlaying()
+{
+	if (mCurrentInternetStreamp)
+	{
+		return 1; // Active and playing
+	}
+	else if (!mURL.empty())
+	{
+		return 2; // "Paused"
+	}
+	else
+	{
+		return 0;
+	}
+}
+
+
+F32 LLStreamingAudio_FMOD::getGain()
+{
+	return mGain;
+}
+
+
+std::string LLStreamingAudio_FMOD::getURL()
+{
+	return mURL;
+}
+
+
+void LLStreamingAudio_FMOD::setGain(F32 vol)
+{
+	mGain = vol;
+
+	if (mFMODInternetStreamChannel != -1)
+	{
+		vol = llclamp(vol, 0.f, 1.f);
+		int vol_int = llround(vol * 255.f);
+		FSOUND_SetVolumeAbsolute(mFMODInternetStreamChannel, vol_int);
+	}
+}
+
+
+///////////////////////////////////////////////////////
+// manager of possibly-multiple internet audio streams
+
+LLAudioStreamManagerFMOD::LLAudioStreamManagerFMOD(const std::string& url) :
+	mInternetStream(NULL),
+	mReady(false)
+{
+	mInternetStreamURL = url;
+	mInternetStream = FSOUND_Stream_Open(url.c_str(), FSOUND_NORMAL | FSOUND_NONBLOCKING, 0, 0);
+	if (!mInternetStream)
+	{
+		llwarns << "Couldn't open fmod stream, error "
+			<< FMOD_ErrorString(FSOUND_GetError())
+			<< llendl;
+		mReady = false;
+		return;
+	}
+
+	mReady = true;
+}
+
+int LLAudioStreamManagerFMOD::startStream()
+{
+	// We need a live and opened stream before we try and play it.
+	if (!mInternetStream || getOpenState())
+	{
+		llwarns << "No internet stream to start playing!" << llendl;
+		return -1;
+	}
+
+	// Make sure the stream is set to 2D mode.
+	FSOUND_Stream_SetMode(mInternetStream, FSOUND_2D);
+
+	return FSOUND_Stream_PlayEx(FSOUND_FREE, mInternetStream, NULL, true);
+}
+
+bool LLAudioStreamManagerFMOD::stopStream()
+{
+	if (mInternetStream)
+	{
+		int read_percent = 0;
+		int status = 0;
+		int bitrate = 0;
+		unsigned int flags = 0x0;
+		FSOUND_Stream_Net_GetStatus(mInternetStream, &status, &read_percent, &bitrate, &flags);
+
+		bool close = true;
+		switch (status)
+		{
+		case FSOUND_STREAM_NET_CONNECTING:
+			close = false;
+			break;
+		case FSOUND_STREAM_NET_NOTCONNECTED:
+		case FSOUND_STREAM_NET_BUFFERING:
+		case FSOUND_STREAM_NET_READY:
+		case FSOUND_STREAM_NET_ERROR:
+		default:
+			close = true;
+		}
+
+		if (close)
+		{
+			FSOUND_Stream_Close(mInternetStream);
+			mInternetStream = NULL;
+			return true;
+		}
+		else
+		{
+			return false;
+		}
+	}
+	else
+	{
+		return true;
+	}
+}
+
+int LLAudioStreamManagerFMOD::getOpenState()
+{
+	int open_state = FSOUND_Stream_GetOpenState(mInternetStream);
+	return open_state;
+}
diff --git a/indra/llaudio/llstreamingaudio_fmod.h b/indra/llaudio/llstreamingaudio_fmod.h
new file mode 100644
index 0000000000..968ab53a0b
--- /dev/null
+++ b/indra/llaudio/llstreamingaudio_fmod.h
@@ -0,0 +1,68 @@
+/** 
+ * @file streamingaudio_fmod.h
+ * @author Tofu Linden
+ * @brief Definition of LLStreamingAudio_FMOD implementation
+ *
+ * $LicenseInfo:firstyear=2009&license=viewergpl$
+ * 
+ * Copyright (c) 2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#ifndef LL_STREAMINGAUDIO_FMOD_H
+#define LL_STREAMINGAUDIO_FMOD_H
+
+#include "stdtypes.h" // from llcommon
+
+#include "llstreamingaudio.h"
+
+class LLAudioStreamManagerFMOD;
+
+class LLStreamingAudio_FMOD : public LLStreamingAudioInterface
+{
+ public:
+	LLStreamingAudio_FMOD();
+	/*virtual*/ ~LLStreamingAudio_FMOD();
+
+	/*virtual*/ void start(const std::string& url);
+	/*virtual*/ void stop();
+	/*virtual*/ void pause(int pause);
+	/*virtual*/ void update();
+	/*virtual*/ int isPlaying();
+	/*virtual*/ void setGain(F32 vol);
+	/*virtual*/ F32 getGain();
+	/*virtual*/ std::string getURL();
+
+private:
+	LLAudioStreamManagerFMOD *mCurrentInternetStreamp;
+	int mFMODInternetStreamChannel;
+	std::list<LLAudioStreamManagerFMOD *> mDeadStreams;
+
+	std::string mURL;
+	F32 mGain;
+};
+
+
+#endif // LL_STREAMINGAUDIO_FMOD_H
diff --git a/indra/llaudio/llvorbisencode.cpp b/indra/llaudio/llvorbisencode.cpp
new file mode 100644
index 0000000000..8ee082a245
--- /dev/null
+++ b/indra/llaudio/llvorbisencode.cpp
@@ -0,0 +1,505 @@
+/** 
+ * @file vorbisencode.cpp
+ * @brief Vorbis encoding routine routine for Indra.
+ *
+ * $LicenseInfo:firstyear=2000&license=viewergpl$
+ * 
+ * Copyright (c) 2000-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#include "linden_common.h"
+
+#include "vorbis/vorbisenc.h"
+
+#include "llvorbisencode.h"
+#include "llerror.h"
+#include "llrand.h"
+#include "llmath.h"
+#include "llapr.h"
+
+//#if LL_DARWIN
+// MBW -- XXX -- Getting rid of SecondLifeVorbis for now -- no fmod means no name collisions.
+#if 0
+#include "VorbisFramework.h"
+
+#define vorbis_analysis				mac_vorbis_analysis
+#define vorbis_analysis_headerout	mac_vorbis_analysis_headerout
+#define vorbis_analysis_init		mac_vorbis_analysis_init
+#define vorbis_encode_ctl			mac_vorbis_encode_ctl
+#define vorbis_encode_setup_init	mac_vorbis_encode_setup_init
+#define vorbis_encode_setup_managed	mac_vorbis_encode_setup_managed
+
+#define	vorbis_info_init			mac_vorbis_info_init
+#define	vorbis_info_clear			mac_vorbis_info_clear
+#define	vorbis_comment_init			mac_vorbis_comment_init
+#define	vorbis_comment_clear		mac_vorbis_comment_clear
+#define	vorbis_block_init			mac_vorbis_block_init
+#define	vorbis_block_clear			mac_vorbis_block_clear
+#define	vorbis_dsp_clear			mac_vorbis_dsp_clear
+#define	vorbis_analysis_buffer		mac_vorbis_analysis_buffer
+#define	vorbis_analysis_wrote		mac_vorbis_analysis_wrote
+#define	vorbis_analysis_blockout	mac_vorbis_analysis_blockout
+
+#define ogg_stream_packetin			mac_ogg_stream_packetin
+#define	ogg_stream_init				mac_ogg_stream_init
+#define	ogg_stream_flush			mac_ogg_stream_flush
+#define	ogg_stream_pageout			mac_ogg_stream_pageout
+#define	ogg_page_eos				mac_ogg_page_eos
+#define	ogg_stream_clear			mac_ogg_stream_clear
+
+#endif
+
+S32 check_for_invalid_wav_formats(const std::string& in_fname, std::string& error_msg)
+{
+	U16 num_channels = 0;
+	U32 sample_rate = 0;
+	U32 bits_per_sample = 0;
+	U32 physical_file_size = 0;
+	U32 chunk_length = 0;
+	U32 raw_data_length = 0;
+	U32 bytes_per_sec = 0;
+	BOOL uncompressed_pcm = FALSE;
+
+	unsigned char wav_header[44];		/*Flawfinder: ignore*/
+
+	error_msg.clear();
+
+	//********************************
+	LLAPRFile infile ;
+    infile.open(in_fname,LL_APR_RB);
+	//********************************
+	if (!infile.getFileHandle())
+	{
+		error_msg = "CannotUploadSoundFile";
+		return(LLVORBISENC_SOURCE_OPEN_ERR);
+	}
+
+	infile.read(wav_header, 44);
+	physical_file_size = infile.seek(APR_END,0);
+
+	if (strncmp((char *)&(wav_header[0]),"RIFF",4))
+	{
+		error_msg = "SoundFileNotRIFF";
+		return(LLVORBISENC_WAV_FORMAT_ERR);
+	}
+
+	if (strncmp((char *)&(wav_header[8]),"WAVE",4))
+	{
+		error_msg = "SoundFileNotRIFF";
+		return(LLVORBISENC_WAV_FORMAT_ERR);
+	}
+	
+	// parse the chunks
+	
+	U32 file_pos = 12;  // start at the first chunk (usually fmt but not always)
+	
+	while ((file_pos + 8)< physical_file_size)
+	{
+		infile.seek(APR_SET,file_pos);
+		infile.read(wav_header, 44);
+
+		chunk_length = ((U32) wav_header[7] << 24) 
+			+ ((U32) wav_header[6] << 16) 
+			+ ((U32) wav_header[5] << 8) 
+			+ wav_header[4];
+
+//		llinfos << "chunk found: '" << wav_header[0] << wav_header[1] << wav_header[2] << wav_header[3] << "'" << llendl;
+
+		if (!(strncmp((char *)&(wav_header[0]),"fmt ",4)))
+		{
+			if ((wav_header[8] == 0x01) && (wav_header[9] == 0x00))
+			{
+				uncompressed_pcm = TRUE;
+			}
+			num_channels = ((U16) wav_header[11] << 8) + wav_header[10];
+			sample_rate = ((U32) wav_header[15] << 24) 
+				+ ((U32) wav_header[14] << 16) 
+				+ ((U32) wav_header[13] << 8) 
+				+ wav_header[12];
+			bits_per_sample = ((U16) wav_header[23] << 8) + wav_header[22];
+			bytes_per_sec = ((U32) wav_header[19] << 24) 
+				+ ((U32) wav_header[18] << 16) 
+				+ ((U32) wav_header[17] << 8) 
+				+ wav_header[16];
+		}
+		else if (!(strncmp((char *)&(wav_header[0]),"data",4)))
+		{
+			raw_data_length = chunk_length;			
+		}
+		file_pos += (chunk_length + 8);
+		chunk_length = 0;
+	} 
+	//****************
+	infile.close();
+	//****************
+
+	if (!uncompressed_pcm)
+	{	
+		 error_msg = "SoundFileNotPCM";
+		  return(LLVORBISENC_PCM_FORMAT_ERR);
+	}
+	
+	if ((num_channels < 1) || (num_channels > 2))
+	{	
+		error_msg = "SoundFileInvalidChannelCount";
+		return(LLVORBISENC_MULTICHANNEL_ERR);
+	}
+
+	if (sample_rate != 44100)
+	{	
+		error_msg = "SoundFileInvalidSampleRate";
+		return(LLVORBISENC_UNSUPPORTED_SAMPLE_RATE);
+	}
+	
+	if ((bits_per_sample != 16) && (bits_per_sample != 8))
+	{		 
+		error_msg = "SoundFileInvalidWordSize";
+		return(LLVORBISENC_UNSUPPORTED_WORD_SIZE);
+	}
+
+	if (!raw_data_length)
+	{
+		error_msg = "SoundFileInvalidHeader";
+		return(LLVORBISENC_CLIP_TOO_LONG);		 
+	}
+
+	F32 clip_length = (F32)raw_data_length/(F32)bytes_per_sec;
+		
+	if (clip_length > 10.0f)
+	{
+		error_msg = "SoundFileInvalidTooLong";
+		return(LLVORBISENC_CLIP_TOO_LONG);		 
+	}
+
+    return(LLVORBISENC_NOERR);
+}
+
+S32 encode_vorbis_file(const std::string& in_fname, const std::string& out_fname)
+{
+#define READ_BUFFER 1024
+	unsigned char readbuffer[READ_BUFFER*4+44];   /* out of the data segment, not the stack */	/*Flawfinder: ignore*/
+
+	ogg_stream_state os; /* take physical pages, weld into a logical stream of packets */
+	ogg_page         og; /* one Ogg bitstream page.  Vorbis packets are inside */
+	ogg_packet       op; /* one raw packet of data for decode */
+	
+	vorbis_info      vi; /* struct that stores all the static vorbis bitstream settings */
+	vorbis_comment   vc; /* struct that stores all the user comments */
+	
+	vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
+	vorbis_block     vb; /* local working space for packet->PCM decode */
+	
+	int eos=0;
+	int result;
+
+	U16 num_channels = 0;
+	U32 sample_rate = 0;
+	U32 bits_per_sample = 0;
+
+	S32 format_error = 0;
+	std::string error_msg;
+	if ((format_error = check_for_invalid_wav_formats(in_fname, error_msg)))
+	{
+		llwarns << error_msg << ": " << in_fname << llendl;
+		return(format_error);
+	}
+
+#if 1
+	unsigned char wav_header[44];	/*Flawfinder: ignore*/
+
+	S32 data_left = 0;
+
+	LLAPRFile infile ;
+	infile.open(in_fname,LL_APR_RB);
+	if (!infile.getFileHandle())
+	{
+		llwarns << "Couldn't open temporary ogg file for writing: " << in_fname
+			<< llendl;
+		return(LLVORBISENC_SOURCE_OPEN_ERR);
+	}
+
+	LLAPRFile outfile ;
+	outfile.open(out_fname,LL_APR_WPB);
+	if (!outfile.getFileHandle())
+	{
+		llwarns << "Couldn't open upload sound file for reading: " << in_fname
+			<< llendl;
+		return(LLVORBISENC_DEST_OPEN_ERR);
+	}
+	
+	 // parse the chunks
+	 U32 chunk_length = 0;
+	 U32 file_pos = 12;  // start at the first chunk (usually fmt but not always)
+	 
+	 while (infile.eof() != APR_EOF)
+	 {
+		 infile.seek(APR_SET,file_pos);
+		 infile.read(wav_header, 44);
+		 
+		 chunk_length = ((U32) wav_header[7] << 24) 
+			 + ((U32) wav_header[6] << 16) 
+			 + ((U32) wav_header[5] << 8) 
+			 + wav_header[4];
+		 
+//		 llinfos << "chunk found: '" << wav_header[0] << wav_header[1] << wav_header[2] << wav_header[3] << "'" << llendl;
+		 
+		 if (!(strncmp((char *)&(wav_header[0]),"fmt ",4)))
+		 {
+			 num_channels = ((U16) wav_header[11] << 8) + wav_header[10];
+			 sample_rate = ((U32) wav_header[15] << 24) 
+				 + ((U32) wav_header[14] << 16) 
+				 + ((U32) wav_header[13] << 8) 
+				 + wav_header[12];
+			 bits_per_sample = ((U16) wav_header[23] << 8) + wav_header[22];
+		 }
+	 	 else if (!(strncmp((char *)&(wav_header[0]),"data",4)))
+		 {
+			 infile.seek(APR_SET,file_pos+8);
+			 // leave the file pointer at the beginning of the data chunk data
+			 data_left = chunk_length;			
+			 break;
+		 }
+		 file_pos += (chunk_length + 8);
+		 chunk_length = 0;
+	 } 
+	 
+
+	 /********** Encode setup ************/
+	 
+	 /* choose an encoding mode */
+	 /* (mode 0: 44kHz stereo uncoupled, roughly 128kbps VBR) */
+	 vorbis_info_init(&vi);
+
+	 // always encode to mono
+
+	 // SL-52913 & SL-53779 determined this quality level to be our 'good
+	 // enough' general-purpose quality level with a nice low bitrate.
+	 // Equivalent to oggenc -q0.5
+	 F32 quality = 0.05f;
+//	 quality = (bitrate==128000 ? 0.4f : 0.1);
+
+//	 if (vorbis_encode_init(&vi, /* num_channels */ 1 ,sample_rate, -1, bitrate, -1))
+	 if (vorbis_encode_init_vbr(&vi, /* num_channels */ 1 ,sample_rate, quality))
+//	 if (vorbis_encode_setup_managed(&vi,1,sample_rate,-1,bitrate,-1) ||
+//		vorbis_encode_ctl(&vi,OV_ECTL_RATEMANAGE_AVG,NULL) ||
+//		vorbis_encode_setup_init(&vi))
+	{
+		llwarns << "unable to initialize vorbis codec at quality " << quality << llendl;
+		//		llwarns << "unable to initialize vorbis codec at bitrate " << bitrate << llendl;
+		return(LLVORBISENC_DEST_OPEN_ERR);
+	}
+	 
+	 /* add a comment */
+	 vorbis_comment_init(&vc);
+//	 vorbis_comment_add(&vc,"Linden");
+	 
+	 /* set up the analysis state and auxiliary encoding storage */
+	 vorbis_analysis_init(&vd,&vi);
+	 vorbis_block_init(&vd,&vb);
+	 
+	 /* set up our packet->stream encoder */
+	 /* pick a random serial number; that way we can more likely build
+		chained streams just by concatenation */
+	 ogg_stream_init(&os, ll_rand());
+	 
+	 /* Vorbis streams begin with three headers; the initial header (with
+		most of the codec setup parameters) which is mandated by the Ogg
+		bitstream spec.  The second header holds any comment fields.  The
+		third header holds the bitstream codebook.  We merely need to
+		make the headers, then pass them to libvorbis one at a time;
+		libvorbis handles the additional Ogg bitstream constraints */
+	 
+	 {
+		 ogg_packet header;
+		 ogg_packet header_comm;
+		 ogg_packet header_code;
+		 
+		 vorbis_analysis_headerout(&vd,&vc,&header,&header_comm,&header_code);
+		 ogg_stream_packetin(&os,&header); /* automatically placed in its own
+											  page */
+		 ogg_stream_packetin(&os,&header_comm);
+		 ogg_stream_packetin(&os,&header_code);
+		 
+		 /* We don't have to write out here, but doing so makes streaming 
+		  * much easier, so we do, flushing ALL pages. This ensures the actual
+		  * audio data will start on a new page
+		  */
+		 while(!eos){
+			 int result=ogg_stream_flush(&os,&og);
+			 if(result==0)break;
+			 outfile.write(og.header, og.header_len);
+			 outfile.write(og.body, og.body_len);
+		 }
+		 
+	 }
+	 
+	 
+	 while(!eos)
+	 {
+		 long bytes_per_sample = bits_per_sample/8;
+
+		 long bytes=(long)infile.read(readbuffer,llclamp((S32)(READ_BUFFER*num_channels*bytes_per_sample),0,data_left)); /* stereo hardwired here */
+		 
+		 if (bytes==0)
+		 {
+			 /* end of file.  this can be done implicitly in the mainline,
+				but it's easier to see here in non-clever fashion.
+				Tell the library we're at end of stream so that it can handle
+				the last frame and mark end of stream in the output properly */
+
+			 vorbis_analysis_wrote(&vd,0);
+//			 eos = 1;
+			 
+		 }
+		 else
+		 {
+			 long i;
+			 long samples;
+			 int temp;
+
+			 data_left -= bytes;
+             /* data to encode */
+			 
+			 /* expose the buffer to submit data */
+			 float **buffer=vorbis_analysis_buffer(&vd,READ_BUFFER);
+			
+			 i = 0;
+			 samples = bytes / (num_channels * bytes_per_sample);
+
+			 if (num_channels == 2)
+			 {
+				 if (bytes_per_sample == 2)
+				 {
+					 /* uninterleave samples */
+					 for(i=0; i<samples ;i++)
+					 {
+					 	 temp =  ((signed char *)readbuffer)[i*4+1];	/*Flawfinder: ignore*/
+						 temp += ((signed char *)readbuffer)[i*4+3];	/*Flawfinder: ignore*/
+						 temp <<= 8;
+						 temp += readbuffer[i*4];
+						 temp += readbuffer[i*4+2];
+
+						 buffer[0][i] = ((float)temp) / 65536.f;
+					 }
+				 }
+				 else // presume it's 1 byte per which is unsigned (F#@%ing wav "standard")
+				 {
+					 /* uninterleave samples */
+					 for(i=0; i<samples ;i++)
+					 {
+					 	 temp  = readbuffer[i*2+0];
+						 temp += readbuffer[i*2+1];
+						 temp -= 256;
+						 buffer[0][i] = ((float)temp) / 256.f;
+					 }
+				 } 
+			 }
+			 else if (num_channels == 1)
+			 {
+				 if (bytes_per_sample == 2)
+				 {
+					 for(i=0; i < samples ;i++)
+					 {
+					 	 temp = ((signed char*)readbuffer)[i*2+1];
+						 temp <<= 8;
+						 temp += readbuffer[i*2];
+						 buffer[0][i] = ((float)temp) / 32768.f;
+					 }
+				 }
+				 else // presume it's 1 byte per which is unsigned (F#@%ing wav "standard")
+				 {
+					 for(i=0; i < samples ;i++)
+					 {
+						 temp = readbuffer[i];
+						 temp -= 128;
+						 buffer[0][i] = ((float)temp) / 128.f;
+					 }
+				 }
+			 }
+				
+			 /* tell the library how much we actually submitted */
+			 vorbis_analysis_wrote(&vd,i);
+		 }
+			 
+		 /* vorbis does some data preanalysis, then divvies up blocks for
+			more involved (potentially parallel) processing.  Get a single
+			block for encoding now */
+		 while(vorbis_analysis_blockout(&vd,&vb)==1)
+		 {
+			 
+			 /* analysis */
+			/* Do the main analysis, creating a packet */
+			vorbis_analysis(&vb, NULL);
+			vorbis_bitrate_addblock(&vb);
+
+			while(vorbis_bitrate_flushpacket(&vd, &op)) 
+			{
+			 
+			 /* weld the packet into the bitstream */
+			 ogg_stream_packetin(&os,&op);
+			 
+			 /* write out pages (if any) */
+			 while(!eos)
+			 {
+				 result = ogg_stream_pageout(&os,&og);
+
+				 if(result==0)
+				 	break;
+
+				 outfile.write(og.header, og.header_len);
+				 outfile.write(og.body, og.body_len);
+				 
+				 /* this could be set above, but for illustrative purposes, I do
+					it here (to show that vorbis does know where the stream ends) */
+				 
+				 if(ogg_page_eos(&og))
+				 	eos=1;
+				 
+			 }
+			}
+		 }
+	 }
+	 
+	 
+	 
+	 /* clean up and exit.  vorbis_info_clear() must be called last */
+	 
+	 ogg_stream_clear(&os);
+	 vorbis_block_clear(&vb);
+	 vorbis_dsp_clear(&vd);
+	 vorbis_comment_clear(&vc);
+	 vorbis_info_clear(&vi);
+	 
+	 /* ogg_page and ogg_packet structs always point to storage in
+		libvorbis.  They're never freed or manipulated directly */
+	 
+//	 fprintf(stderr,"Vorbis encoding: Done.\n");
+	 llinfos << "Vorbis encoding: Done." << llendl;
+	 
+#endif
+	 return(LLVORBISENC_NOERR);
+	 
+}
diff --git a/indra/llaudio/llvorbisencode.h b/indra/llaudio/llvorbisencode.h
new file mode 100644
index 0000000000..ff5ce3a053
--- /dev/null
+++ b/indra/llaudio/llvorbisencode.h
@@ -0,0 +1,53 @@
+/** 
+ * @file vorbisencode.h
+ * @brief Vorbis encoding routine routine for Indra.
+ *
+ * $LicenseInfo:firstyear=2000&license=viewergpl$
+ * 
+ * Copyright (c) 2000-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#ifndef LL_VORBISENCODE_H
+#define LL_VORBISENCODE_H
+
+const S32 LLVORBISENC_NOERR                        = 0; // no error
+const S32 LLVORBISENC_SOURCE_OPEN_ERR              = 1; // error opening source
+const S32 LLVORBISENC_DEST_OPEN_ERR                = 2; // error opening destination
+const S32 LLVORBISENC_WAV_FORMAT_ERR               = 3; // not a WAV
+const S32 LLVORBISENC_PCM_FORMAT_ERR               = 4; // not a PCM
+const S32 LLVORBISENC_MONO_ERR                     = 5; // can't do mono
+const S32 LLVORBISENC_STEREO_ERR                   = 6; // can't do stereo
+const S32 LLVORBISENC_MULTICHANNEL_ERR             = 7; // can't do stereo
+const S32 LLVORBISENC_UNSUPPORTED_SAMPLE_RATE      = 8; // unsupported sample rate
+const S32 LLVORBISENC_UNSUPPORTED_WORD_SIZE        = 9; // unsupported word size
+const S32 LLVORBISENC_CLIP_TOO_LONG                = 10; // source file is too long
+
+
+S32 check_for_invalid_wav_formats(const std::string& in_fname, std::string& error_msg);
+S32 encode_vorbis_file(const std::string& in_fname, const std::string& out_fname);
+
+#endif
+
diff --git a/indra/llaudio/llwindgen.h b/indra/llaudio/llwindgen.h
new file mode 100644
index 0000000000..847bfa6e9d
--- /dev/null
+++ b/indra/llaudio/llwindgen.h
@@ -0,0 +1,136 @@
+/** 
+ * @file windgen.h
+ * @brief Templated wind noise generation
+ *
+ * $LicenseInfo:firstyear=2002&license=viewergpl$
+ * 
+ * Copyright (c) 2002-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+#ifndef WINDGEN_H
+#define WINDGEN_H
+
+#include "llcommon.h"
+#include "llrand.h"
+
+template <class MIXBUFFERFORMAT_T>
+class LLWindGen
+{
+public:
+	LLWindGen() :
+		mTargetGain(0.f),
+		mTargetFreq(100.f),
+		mTargetPanGainR(0.5f),
+		mbuf0(0.0),
+		mbuf1(0.0),
+		mbuf2(0.0),
+		mbuf3(0.0),
+		mbuf4(0.0),
+		mbuf5(0.0),
+		mY0(0.0),
+		mY1(0.0),
+		mCurrentGain(0.f),
+		mCurrentFreq(100.f),
+		mCurrentPanGainR(0.5f) {};
+
+	static const U32 getInputSamplingRate() {return mInputSamplingRate;}
+
+	// newbuffer = the buffer passed from the previous DSP unit.
+	// numsamples = length in samples-per-channel at this mix time.
+	// stride = number of bytes between start of each sample.
+	// NOTE: generates L/R interleaved stereo
+	MIXBUFFERFORMAT_T* windGenerate(MIXBUFFERFORMAT_T *newbuffer, int numsamples, int stride)
+	{
+		U8 *cursamplep = (U8*)newbuffer;
+		
+		double bandwidth = 50.0F;
+		double a0,b1,b2;
+		
+		// calculate resonant filter coeffs
+		b2 = exp(-(F_TWO_PI) * (bandwidth / mInputSamplingRate));
+		
+		while (numsamples--)
+		{
+			mCurrentFreq = (float)((0.999 * mCurrentFreq) + (0.001 * mTargetFreq));
+			mCurrentGain = (float)((0.999 * mCurrentGain) + (0.001 * mTargetGain));
+			mCurrentPanGainR = (float)((0.999 * mCurrentPanGainR) + (0.001 * mTargetPanGainR));
+			b1 = (-4.0 * b2) / (1.0 + b2) * cos(F_TWO_PI * (mCurrentFreq / mInputSamplingRate));
+			a0 = (1.0 - b2) * sqrt(1.0 - (b1 * b1) / (4.0 * b2));
+			double nextSample;
+			
+			// start with white noise
+			nextSample = ll_frand(2.0f) - 1.0f;
+			
+			// apply pinking filter
+			mbuf0 = 0.997f * mbuf0 + 0.0126502f * nextSample; 
+			mbuf1 = 0.985f * mbuf1 + 0.0139083f * nextSample;
+			mbuf2 = 0.950f * mbuf2 + 0.0205439f * nextSample;
+			mbuf3 = 0.850f * mbuf3 + 0.0387225f * nextSample;
+			mbuf4 = 0.620f * mbuf4 + 0.0465932f * nextSample;
+			mbuf5 = 0.250f * mbuf5 + 0.1093477f * nextSample;
+			
+			nextSample = mbuf0 + mbuf1 + mbuf2 + mbuf3 + mbuf4 + mbuf5;
+			
+			// do a resonant filter on the noise
+			nextSample = (double)( a0 * nextSample - b1 * mY0 - b2 * mY1 );
+			mY1 = mY0;
+			mY0 = nextSample;
+			
+			nextSample *= mCurrentGain;
+			
+			MIXBUFFERFORMAT_T	sample;
+			
+			sample = llfloor(((F32)nextSample*32768.f*(1.0f - mCurrentPanGainR))+0.5f);
+			*(MIXBUFFERFORMAT_T*)cursamplep = llclamp(sample, (MIXBUFFERFORMAT_T)-32768, (MIXBUFFERFORMAT_T)32767);
+			cursamplep += stride;
+
+			sample = llfloor(((F32)nextSample*32768.f*mCurrentPanGainR)+0.5f);
+			*(MIXBUFFERFORMAT_T*)cursamplep = llclamp(sample, (MIXBUFFERFORMAT_T)-32768, (MIXBUFFERFORMAT_T)32767);
+			cursamplep += stride;
+		}
+		
+		return newbuffer;
+	}
+
+	F32 mTargetGain;
+	F32 mTargetFreq;
+	F32 mTargetPanGainR;
+
+private:
+	static const U32 mInputSamplingRate = 44100;
+	F64 mbuf0;
+	F64 mbuf1;
+	F64 mbuf2;
+	F64 mbuf3;
+	F64 mbuf4;
+	F64 mbuf5;
+	F64 mY0;
+	F64 mY1;
+	F32 mCurrentGain;
+	F32 mCurrentFreq;
+	F32 mCurrentPanGainR;
+};
+
+#endif