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working on selected audio

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This commit is contained in:
ZappoMan 2013-07-22 23:05:42 -07:00
parent c301b799c5
commit c969570e8c
8 changed files with 233 additions and 145 deletions
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32
audio-mixer/src/PositionalAudioRingBuffer.cpp
View file

@ -32,6 +32,32 @@ PositionalAudioRingBuffer::~PositionalAudioRingBuffer() {
}
}
bool PositionalAudioRingBuffer::isListeningToSource(PositionalAudioRingBuffer* other) {
switch (_listenMode) {
default:
case AudioRingBuffer::NORMAL:
return true;
break;
case AudioRingBuffer::OMNI_DIRECTIONAL_POINT: {
float distance = glm::distance(_position, other->_position);
return distance <= _listenRadius;
break;
}
case AudioRingBuffer::SELECTED_SOURCES:
if (_listenSources) {
for (int i = 0; i < _listenSourceCount; i++) {
if (other->_sourceID == _listenSources[i]) {
return true;
}
}
}
return false;
break;
}
}
int PositionalAudioRingBuffer::parseData(unsigned char* sourceBuffer, int numBytes) {
unsigned char* currentBuffer = sourceBuffer + numBytesForPacketHeader(sourceBuffer);
currentBuffer += parseSourceData(currentBuffer, numBytes - (currentBuffer - sourceBuffer));
@ -56,21 +82,19 @@ int PositionalAudioRingBuffer::parseListenModeData(unsigned char* sourceBuffer,
memcpy(&_listenMode, currentBuffer, sizeof(_listenMode));
currentBuffer += sizeof(_listenMode);
if (_listenMode == AudioRingBuffer::OMNI_DIRECTIONAL_POINT) {
memcpy(&_listenRadius, currentBuffer, sizeof(_listenRadius));
currentBuffer += sizeof(_listenRadius);
} else if (_listenMode == AudioRingBuffer::SELECTED_SOURCES) {
memcpy(&_listenSourceCount, currentBuffer, sizeof(_listenSourceCount));
currentBuffer += sizeof(_listenSourceCount);
if (_listenSources) {
delete[] _listenSources;
}
_listenSources = new int[_listenSourceCount];
memcpy(_listenSources, currentBuffer, sizeof(int) * _listenSourceCount);
currentBuffer += sizeof(int) * _listenSourceCount;
}
return currentBuffer - sourceBuffer;
@ -81,7 +105,7 @@ int PositionalAudioRingBuffer::parsePositionalData(unsigned char* sourceBuffer,
memcpy(&_position, currentBuffer, sizeof(_position));
currentBuffer += sizeof(_position);
memcpy(&_orientation, currentBuffer, sizeof(_orientation));
currentBuffer += sizeof(_orientation);

4
audio-mixer/src/PositionalAudioRingBuffer.h
View file

@ -30,6 +30,10 @@ public:
const glm::vec3& getPosition() const { return _position; }
const glm::quat& getOrientation() const { return _orientation; }
bool isListeningToSource(PositionalAudioRingBuffer* other);
int getSourceID() const { return _sourceID; }
int getListeningMode() const { return _listenMode; }
protected:
// disallow copying of PositionalAudioRingBuffer objects

281
audio-mixer/src/main.cpp
View file

@ -159,168 +159,174 @@ int main(int argc, const char* argv[]) {
// zero out the client mix for this node
memset(clientSamples, 0, sizeof(clientSamples));
// loop through all other nodes that have sufficient audio to mix
for (NodeList::iterator otherNode = nodeList->begin(); otherNode != nodeList->end(); otherNode++) {
if (((PositionalAudioRingBuffer*) otherNode->getLinkedData())->willBeAddedToMix()
&& (otherNode != node || (otherNode == node && nodeRingBuffer->shouldLoopbackForNode()))) {
PositionalAudioRingBuffer* otherNodeBuffer = (PositionalAudioRingBuffer*) otherNode->getLinkedData();
// based on our listen mode we will do this mixing...
if (nodeRingBuffer->isListeningToSource(otherNodeBuffer)) {
float bearingRelativeAngleToSource = 0.0f;
float attenuationCoefficient = 1.0f;
int numSamplesDelay = 0;
float weakChannelAmplitudeRatio = 1.0f;
float bearingRelativeAngleToSource = 0.0f;
float attenuationCoefficient = 1.0f;
int numSamplesDelay = 0;
float weakChannelAmplitudeRatio = 1.0f;
stk::TwoPole* otherNodeTwoPole = NULL;
stk::TwoPole* otherNodeTwoPole = NULL;
if (otherNode != node) {
// only do axis/distance attenuation when in normal mode
if (otherNode != node && nodeRingBuffer->getListeningMode() == AudioRingBuffer::NORMAL) {
glm::vec3 listenerPosition = nodeRingBuffer->getPosition();
glm::vec3 relativePosition = otherNodeBuffer->getPosition() - nodeRingBuffer->getPosition();
glm::quat inverseOrientation = glm::inverse(nodeRingBuffer->getOrientation());
glm::vec3 listenerPosition = nodeRingBuffer->getPosition();
glm::vec3 relativePosition = otherNodeBuffer->getPosition() - nodeRingBuffer->getPosition();
glm::quat inverseOrientation = glm::inverse(nodeRingBuffer->getOrientation());
float distanceSquareToSource = glm::dot(relativePosition, relativePosition);
float radius = 0.0f;
float distanceSquareToSource = glm::dot(relativePosition, relativePosition);
float radius = 0.0f;
if (otherNode->getType() == NODE_TYPE_AUDIO_INJECTOR) {
InjectedAudioRingBuffer* injectedBuffer = (InjectedAudioRingBuffer*) otherNodeBuffer;
radius = injectedBuffer->getRadius();
attenuationCoefficient *= injectedBuffer->getAttenuationRatio();
}
if (otherNode->getType() == NODE_TYPE_AUDIO_INJECTOR) {
InjectedAudioRingBuffer* injectedBuffer = (InjectedAudioRingBuffer*) otherNodeBuffer;
radius = injectedBuffer->getRadius();
attenuationCoefficient *= injectedBuffer->getAttenuationRatio();
}
if (radius == 0 || (distanceSquareToSource > radius * radius)) {
// this is either not a spherical source, or the listener is outside the sphere
if (radius == 0 || (distanceSquareToSource > radius * radius)) {
// this is either not a spherical source, or the listener is outside the sphere
if (radius > 0) {
// this is a spherical source - the distance used for the coefficient
// needs to be the closest point on the boundary to the source
if (radius > 0) {
// this is a spherical source - the distance used for the coefficient
// needs to be the closest point on the boundary to the source
// ovveride the distance to the node with the distance to the point on the
// boundary of the sphere
distanceSquareToSource -= (radius * radius);
// ovveride the distance to the node with the distance to the point on the
// boundary of the sphere
distanceSquareToSource -= (radius * radius);
} else {
// calculate the angle delivery for off-axis attenuation
glm::vec3 rotatedListenerPosition = glm::inverse(otherNodeBuffer->getOrientation())
* relativePosition;
} else {
// calculate the angle delivery for off-axis attenuation
glm::vec3 rotatedListenerPosition = glm::inverse(otherNodeBuffer->getOrientation())
* relativePosition;
float angleOfDelivery = glm::angle(glm::vec3(0.0f, 0.0f, -1.0f),
glm::normalize(rotatedListenerPosition));
float angleOfDelivery = glm::angle(glm::vec3(0.0f, 0.0f, -1.0f),
glm::normalize(rotatedListenerPosition));
const float MAX_OFF_AXIS_ATTENUATION = 0.2f;
const float OFF_AXIS_ATTENUATION_FORMULA_STEP = (1 - MAX_OFF_AXIS_ATTENUATION) / 2.0f;
const float MAX_OFF_AXIS_ATTENUATION = 0.2f;
const float OFF_AXIS_ATTENUATION_FORMULA_STEP = (1 - MAX_OFF_AXIS_ATTENUATION) / 2.0f;
float offAxisCoefficient = MAX_OFF_AXIS_ATTENUATION +
(OFF_AXIS_ATTENUATION_FORMULA_STEP * (angleOfDelivery / 90.0f));
float offAxisCoefficient = MAX_OFF_AXIS_ATTENUATION +
(OFF_AXIS_ATTENUATION_FORMULA_STEP * (angleOfDelivery / 90.0f));
// multiply the current attenuation coefficient by the calculated off axis coefficient
attenuationCoefficient *= offAxisCoefficient;
// multiply the current attenuation coefficient by the calculated off axis coefficient
attenuationCoefficient *= offAxisCoefficient;
}
glm::vec3 rotatedSourcePosition = inverseOrientation * relativePosition;
const float DISTANCE_SCALE = 2.5f;
const float GEOMETRIC_AMPLITUDE_SCALAR = 0.3f;
const float DISTANCE_LOG_BASE = 2.5f;
const float DISTANCE_SCALE_LOG = logf(DISTANCE_SCALE) / logf(DISTANCE_LOG_BASE);
// calculate the distance coefficient using the distance to this node
float distanceCoefficient = powf(GEOMETRIC_AMPLITUDE_SCALAR,
DISTANCE_SCALE_LOG +
(0.5f * logf(distanceSquareToSource) / logf(DISTANCE_LOG_BASE)) - 1);
distanceCoefficient = std::min(1.0f, distanceCoefficient);
// multiply the current attenuation coefficient by the distance coefficient
attenuationCoefficient *= distanceCoefficient;
// project the rotated source position vector onto the XZ plane
rotatedSourcePosition.y = 0.0f;
// produce an oriented angle about the y-axis
bearingRelativeAngleToSource = glm::orientedAngle(glm::vec3(0.0f, 0.0f, -1.0f),
glm::normalize(rotatedSourcePosition),
glm::vec3(0.0f, 1.0f, 0.0f));
const float PHASE_AMPLITUDE_RATIO_AT_90 = 0.5;
// figure out the number of samples of delay and the ratio of the amplitude
// in the weak channel for audio spatialization
float sinRatio = fabsf(sinf(glm::radians(bearingRelativeAngleToSource)));
numSamplesDelay = PHASE_DELAY_AT_90 * sinRatio;
weakChannelAmplitudeRatio = 1 - (PHASE_AMPLITUDE_RATIO_AT_90 * sinRatio);
// grab the TwoPole object for this source, add it if it doesn't exist
TwoPoleNodeMap& nodeTwoPoles = nodeRingBuffer->getTwoPoles();
TwoPoleNodeMap::iterator twoPoleIterator = nodeTwoPoles.find(otherNode->getNodeID());
if (twoPoleIterator == nodeTwoPoles.end()) {
// setup the freeVerb effect for this source for this client
otherNodeTwoPole = nodeTwoPoles[otherNode->getNodeID()] = new stk::TwoPole;
} else {
otherNodeTwoPole = twoPoleIterator->second;
}
// calculate the reasonance for this TwoPole based on angle to source
float TWO_POLE_CUT_OFF_FREQUENCY = 800.0f;
float TWO_POLE_MAX_FILTER_STRENGTH = 0.4f;
otherNodeTwoPole->setResonance(TWO_POLE_CUT_OFF_FREQUENCY,
TWO_POLE_MAX_FILTER_STRENGTH
* fabsf(bearingRelativeAngleToSource) / 180.0f,
true);
}
}
int16_t* sourceBuffer = otherNodeBuffer->getNextOutput();
int16_t* goodChannel = (bearingRelativeAngleToSource > 0.0f)
? clientSamples
: clientSamples + BUFFER_LENGTH_SAMPLES_PER_CHANNEL;
int16_t* delayedChannel = (bearingRelativeAngleToSource > 0.0f)
? clientSamples + BUFFER_LENGTH_SAMPLES_PER_CHANNEL
: clientSamples;
int16_t* delaySamplePointer = otherNodeBuffer->getNextOutput() == otherNodeBuffer->getBuffer()
? otherNodeBuffer->getBuffer() + RING_BUFFER_LENGTH_SAMPLES - numSamplesDelay
: otherNodeBuffer->getNextOutput() - numSamplesDelay;
for (int s = 0; s < BUFFER_LENGTH_SAMPLES_PER_CHANNEL; s++) {
// load up the stkFrameBuffer with this source's samples
stkFrameBuffer[s] = (stk::StkFloat) sourceBuffer[s];
}
// perform the TwoPole effect on the stkFrameBuffer
if (otherNodeTwoPole) {
otherNodeTwoPole->tick(stkFrameBuffer);
}
for (int s = 0; s < BUFFER_LENGTH_SAMPLES_PER_CHANNEL; s++) {
if (s < numSamplesDelay) {
// pull the earlier sample for the delayed channel
int earlierSample = delaySamplePointer[s] * attenuationCoefficient * weakChannelAmplitudeRatio;
glm::vec3 rotatedSourcePosition = inverseOrientation * relativePosition;
const float DISTANCE_SCALE = 2.5f;
const float GEOMETRIC_AMPLITUDE_SCALAR = 0.3f;
const float DISTANCE_LOG_BASE = 2.5f;
const float DISTANCE_SCALE_LOG = logf(DISTANCE_SCALE) / logf(DISTANCE_LOG_BASE);
// calculate the distance coefficient using the distance to this node
float distanceCoefficient = powf(GEOMETRIC_AMPLITUDE_SCALAR,
DISTANCE_SCALE_LOG +
(0.5f * logf(distanceSquareToSource) / logf(DISTANCE_LOG_BASE)) - 1);
distanceCoefficient = std::min(1.0f, distanceCoefficient);
// multiply the current attenuation coefficient by the distance coefficient
attenuationCoefficient *= distanceCoefficient;
// project the rotated source position vector onto the XZ plane
rotatedSourcePosition.y = 0.0f;
// produce an oriented angle about the y-axis
bearingRelativeAngleToSource = glm::orientedAngle(glm::vec3(0.0f, 0.0f, -1.0f),
glm::normalize(rotatedSourcePosition),
glm::vec3(0.0f, 1.0f, 0.0f));
const float PHASE_AMPLITUDE_RATIO_AT_90 = 0.5;
// figure out the number of samples of delay and the ratio of the amplitude
// in the weak channel for audio spatialization
float sinRatio = fabsf(sinf(glm::radians(bearingRelativeAngleToSource)));
numSamplesDelay = PHASE_DELAY_AT_90 * sinRatio;
weakChannelAmplitudeRatio = 1 - (PHASE_AMPLITUDE_RATIO_AT_90 * sinRatio);
// grab the TwoPole object for this source, add it if it doesn't exist
TwoPoleNodeMap& nodeTwoPoles = nodeRingBuffer->getTwoPoles();
TwoPoleNodeMap::iterator twoPoleIterator = nodeTwoPoles.find(otherNode->getNodeID());
if (twoPoleIterator == nodeTwoPoles.end()) {
// setup the freeVerb effect for this source for this client
otherNodeTwoPole = nodeTwoPoles[otherNode->getNodeID()] = new stk::TwoPole;
} else {
otherNodeTwoPole = twoPoleIterator->second;
delayedChannel[s] = glm::clamp(delayedChannel[s] + earlierSample,
MIN_SAMPLE_VALUE,
MAX_SAMPLE_VALUE);
}
// calculate the reasonance for this TwoPole based on angle to source
float TWO_POLE_CUT_OFF_FREQUENCY = 800.0f;
float TWO_POLE_MAX_FILTER_STRENGTH = 0.4f;
otherNodeTwoPole->setResonance(TWO_POLE_CUT_OFF_FREQUENCY,
TWO_POLE_MAX_FILTER_STRENGTH
* fabsf(bearingRelativeAngleToSource) / 180.0f,
true);
}
}
int16_t* sourceBuffer = otherNodeBuffer->getNextOutput();
int16_t* goodChannel = (bearingRelativeAngleToSource > 0.0f)
? clientSamples
: clientSamples + BUFFER_LENGTH_SAMPLES_PER_CHANNEL;
int16_t* delayedChannel = (bearingRelativeAngleToSource > 0.0f)
? clientSamples + BUFFER_LENGTH_SAMPLES_PER_CHANNEL
: clientSamples;
int16_t* delaySamplePointer = otherNodeBuffer->getNextOutput() == otherNodeBuffer->getBuffer()
? otherNodeBuffer->getBuffer() + RING_BUFFER_LENGTH_SAMPLES - numSamplesDelay
: otherNodeBuffer->getNextOutput() - numSamplesDelay;
for (int s = 0; s < BUFFER_LENGTH_SAMPLES_PER_CHANNEL; s++) {
// load up the stkFrameBuffer with this source's samples
stkFrameBuffer[s] = (stk::StkFloat) sourceBuffer[s];
}
// perform the TwoPole effect on the stkFrameBuffer
if (otherNodeTwoPole) {
otherNodeTwoPole->tick(stkFrameBuffer);
}
for (int s = 0; s < BUFFER_LENGTH_SAMPLES_PER_CHANNEL; s++) {
if (s < numSamplesDelay) {
// pull the earlier sample for the delayed channel
int earlierSample = delaySamplePointer[s] * attenuationCoefficient * weakChannelAmplitudeRatio;
delayedChannel[s] = glm::clamp(delayedChannel[s] + earlierSample,
MIN_SAMPLE_VALUE,
MAX_SAMPLE_VALUE);
}
int16_t currentSample = stkFrameBuffer[s] * attenuationCoefficient;
int16_t currentSample = stkFrameBuffer[s] * attenuationCoefficient;
goodChannel[s] = glm::clamp(goodChannel[s] + currentSample,
MIN_SAMPLE_VALUE,
MAX_SAMPLE_VALUE);
goodChannel[s] = glm::clamp(goodChannel[s] + currentSample,
MIN_SAMPLE_VALUE,
MAX_SAMPLE_VALUE);
if (s + numSamplesDelay < BUFFER_LENGTH_SAMPLES_PER_CHANNEL) {
int sumSample = delayedChannel[s + numSamplesDelay]
+ (currentSample * weakChannelAmplitudeRatio);
delayedChannel[s + numSamplesDelay] = glm::clamp(sumSample,
MIN_SAMPLE_VALUE,
MAX_SAMPLE_VALUE);
}
if (s + numSamplesDelay < BUFFER_LENGTH_SAMPLES_PER_CHANNEL) {
int sumSample = delayedChannel[s + numSamplesDelay]
+ (currentSample * weakChannelAmplitudeRatio);
delayedChannel[s + numSamplesDelay] = glm::clamp(sumSample,
MIN_SAMPLE_VALUE,
MAX_SAMPLE_VALUE);
}
if (s >= BUFFER_LENGTH_SAMPLES_PER_CHANNEL - PHASE_DELAY_AT_90) {
// this could be a delayed sample on the next pass
// so store the affected back in the ARB
otherNodeBuffer->getNextOutput()[s] = (int16_t) stkFrameBuffer[s];
if (s >= BUFFER_LENGTH_SAMPLES_PER_CHANNEL - PHASE_DELAY_AT_90) {
// this could be a delayed sample on the next pass
// so store the affected back in the ARB
otherNodeBuffer->getNextOutput()[s] = (int16_t) stkFrameBuffer[s];
}
}
}
}
@ -340,7 +346,6 @@ int main(int argc, const char* argv[]) {
if (nodeBuffer->getNextOutput() >= nodeBuffer->getBuffer() + RING_BUFFER_LENGTH_SAMPLES) {
nodeBuffer->setNextOutput(nodeBuffer->getBuffer());
}
nodeBuffer->setWillBeAddedToMix(false);
}
}

36
interface/src/Application.cpp
View file

@ -1835,6 +1835,11 @@ void Application::initMenu() {
(_simulateLeapHand = debugMenu->addAction("Simulate Leap Hand"))->setCheckable(true);
(_testRaveGlove = debugMenu->addAction("Test RaveGlove"))->setCheckable(true);
QMenu* audioDebugMenu = debugMenu->addMenu("Audio Debugging Tools");
audioDebugMenu->addAction("Listen Mode Normal", this, SLOT(setListenModeNormal()), Qt::CTRL | Qt::Key_1);
audioDebugMenu->addAction("Listen Mode Point/Radius", this, SLOT(setListenModePoint()), Qt::CTRL | Qt::Key_2);
audioDebugMenu->addAction("Listen Mode Single Source", this, SLOT(setListenModeSingleSource()), Qt::CTRL | Qt::Key_3);
QMenu* settingsMenu = menuBar->addMenu("Settings");
(_settingsAutosave = settingsMenu->addAction("Autosave"))->setCheckable(true);
_settingsAutosave->setChecked(true);
@ -1846,6 +1851,37 @@ void Application::initMenu() {
_networkAccessManager = new QNetworkAccessManager(this);
}
void Application::setListenModeNormal() {
_audio.setListenMode(AudioRingBuffer::NORMAL);
}
void Application::setListenModePoint() {
_audio.setListenMode(AudioRingBuffer::OMNI_DIRECTIONAL_POINT);
_audio.setListenRadius(1.0);
}
void Application::setListenModeSingleSource() {
_audio.setListenMode(AudioRingBuffer::SELECTED_SOURCES);
_audio.clearListenSources();
NodeList* nodeList = NodeList::getInstance();
for (NodeList::iterator node = nodeList->begin(); node != nodeList->end(); node++) {
if (node->getLinkedData() != NULL && node->getType() == NODE_TYPE_AGENT) {
Avatar* avatar = (Avatar *) node->getLinkedData();
glm::vec3 headPosition = avatar->getHead().getPosition();
glm::vec3 mouseRayOrigin = _myAvatar.getMouseRayOrigin();
glm::vec3 mouseRayDirection = _myAvatar.getMouseRayDirection();
const float HEAD_SPHERE_RADIUS = 0.07;
if (rayIntersectsSphere(mouseRayOrigin, mouseRayDirection, headPosition, HEAD_SPHERE_RADIUS)) {
int sourceID = avatar->getOwningNode()->getNodeID();
_audio.addListenSource(sourceID);
}
}
}
}
void Application::updateFrustumRenderModeAction() {
switch (_frustumDrawingMode) {
default:

4
interface/src/Application.h
View file

@ -170,6 +170,10 @@ private slots:
void copyVoxels();
void pasteVoxels();
void runTests();
void setListenModeNormal();
void setListenModePoint();
void setListenModeSingleSource();
void renderCoverageMap();
void renderCoverageMapsRecursively(CoverageMap* map);

16
interface/src/Audio.cpp
View file

@ -112,7 +112,7 @@ inline void Audio::performIO(int16_t* inputLeft, int16_t* outputLeft, int16_t* o
// we need the amount of bytes in the buffer + 1 for type
// + 12 for 3 floats for position + float for bearing + 1 attenuation byte
unsigned char dataPacket[BUFFER_LENGTH_BYTES_PER_CHANNEL + leadingBytes];
unsigned char dataPacket[MAX_PACKET_SIZE];
PACKET_TYPE packetType = (Application::getInstance()->shouldEchoAudio())
? PACKET_TYPE_MICROPHONE_AUDIO_WITH_ECHO
@ -123,21 +123,25 @@ inline void Audio::performIO(int16_t* inputLeft, int16_t* outputLeft, int16_t* o
// pack Source Data
memcpy(currentPacketPtr, &_sourceID, sizeof(_sourceID));
currentPacketPtr += (sizeof(_sourceID));
leadingBytes += (sizeof(_sourceID));
// pack Listen Mode Data
memcpy(currentPacketPtr, &_listenMode, sizeof(_listenMode));
currentPacketPtr += (sizeof(_listenMode));
leadingBytes += (sizeof(_listenMode));
if (_listenMode == AudioRingBuffer::OMNI_DIRECTIONAL_POINT) {
memcpy(currentPacketPtr, &_listenRadius, sizeof(_listenRadius));
currentPacketPtr += (sizeof(_listenRadius));
leadingBytes += (sizeof(_listenRadius));
} else if (_listenMode == AudioRingBuffer::SELECTED_SOURCES) {
memcpy(currentPacketPtr, &_listenSourceCount, sizeof(_listenSourceCount));
currentPacketPtr += (sizeof(_listenSourceCount));
leadingBytes += (sizeof(_listenSourceCount));
if (_listenSources) {
memcpy(currentPacketPtr, &_listenSources, sizeof(int) * _listenSourceCount);
memcpy(currentPacketPtr, _listenSources, sizeof(int) * _listenSourceCount);
currentPacketPtr += (sizeof(int) * _listenSourceCount);
leadingBytes += (sizeof(int) * _listenSourceCount);
}
}
@ -356,6 +360,12 @@ void Audio::addListenSource(int sourceID) {
_listenSourceCount++;
}
void Audio::clearListenSources() {
delete[] _listenSources;
_listenSources = NULL;
_listenSourceCount = 0;
}
void Audio::removeListenSource(int sourceID) {
// If we don't yet have a list of listen sources, make one
if (_listenSources) {

1
interface/src/Audio.h
View file

@ -60,6 +60,7 @@ public:
void setListenRadius(float radius) { _listenRadius = radius; };
void addListenSource(int sourceID);
void removeListenSource(int sourceID);
void clearListenSources();
private:
PaStream* _stream;

4
interface/src/avatar/Avatar.h
View file

@ -164,6 +164,10 @@ public:
glm::quat getOrientation () const;
glm::quat getWorldAlignedOrientation() const;
const glm::vec3& getMouseRayOrigin() const { return _mouseRayOrigin; }
const glm::vec3& getMouseRayDirection() const { return _mouseRayDirection; }
glm::vec3 getGravity () const { return _gravity; }
glm::vec3 getUprightHeadPosition() const;