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|
/**
* @file audioengine.cpp
* @brief implementation of LLAudioEngine class abstracting the Open
* AL audio support
*
* $LicenseInfo:firstyear=2000&license=viewerlgpl$
* Second Life Viewer Source Code
* Copyright (C) 2010, Linden Research, Inc.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation;
* version 2.1 of the License only.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
* Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
* $/LicenseInfo$
*/
#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;
// Setting mInternalGain to an out of range value fixes the issue reported in STORM-830.
// There is an edge case in setMasterGain during startup which prevents setInternalGain from
// being called if the master volume setting and mInternalGain both equal 0, so using -1 forces
// the if statement in setMasterGain to execute when the viewer starts up.
mInternalGain = -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
LLAudioEngine::LLAudioPlayState LLAudioEngine::isInternetStreamPlaying()
{
if (mStreamingAudioImpl)
return (LLAudioEngine::LLAudioPlayState) mStreamingAudioImpl->isPlaying();
return LLAudioEngine::AUDIO_STOPPED; // 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.
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->isMuted())
{
++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 || sourcep->isMuted())
{
// Muted, or 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->isMuted())
{
continue;
}
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))
{
mEnableWind = initWind();
}
else if (mEnableWind && (!enable))
{
mEnableWind = false;
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)
{
lldebugs << "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)
{
if (muted != mMuted)
{
mMuted = muted;
setMasterGain(mMasterGain);
}
enableWind(!mMuted);
}
void LLAudioEngine::setMasterGain(const F32 gain)
{
mMasterGain = gain;
F32 internal_gain = getMuted() ? 0.f : gain;
if (internal_gain != mInternalGain)
{
mInternalGain = internal_gain;
setInternalGain(mInternalGain);
}
}
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),
mSourceMuted(false),
mAmbient(false),
mLoop(false),
mSyncMaster(false),
mSyncSlave(false),
mQueueSounds(false),
mPlayedOnce(false),
mType(type),
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 if (isMuted())
{
mPriority = 0.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)
{
// Special abuse of play(); don't play a sound, but kill it.
if (audio_uuid.isNull())
{
if (getChannel())
{
getChannel()->setSource(NULL);
setChannel(NULL);
if (!isMuted())
{
mCurrentDatap = NULL;
}
}
return false;
}
// Reset our age timeout if someone attempts to play the source.
mAgeTimer.reset();
LLAudioData *adp = gAudiop->getAudioData(audio_uuid);
addAudioData(adp);
if (isMuted())
{
return false;
}
bool has_buffer = gAudiop->updateBufferForData(adp, audio_uuid);
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
{
const F32 MAX_AGE = 60.f;
const F32 MAX_UNPLAYED_AGE = 15.f;
const F32 MAX_MUTED_AGE = 11.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 > (mSourceMuted ? MAX_MUTED_AGE : MAX_UNPLAYED_AGE)) || mPlayedOnce)
{
// We don't have a channel assigned, and it's been
// over 15 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 15 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)
{
continue;
}
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;
}
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