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initial changes to allow for multiple buffers per client

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This commit is contained in:
Stephen Birarda 2013-10-21 14:27:55 -07:00
parent 5fa5e7aa78
commit 1b129a43b5
7 changed files with 254 additions and 213 deletions
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409
assignment-client/src/audio/AudioMixer.cpp
View file

@ -48,8 +48,6 @@
#include "AudioMixer.h"
const unsigned short MIXER_LISTEN_PORT = 55443;
const short JITTER_BUFFER_MSECS = 12;
const short JITTER_BUFFER_SAMPLES = JITTER_BUFFER_MSECS * (SAMPLE_RATE / 1000.0);
@ -70,6 +68,201 @@ AudioMixer::AudioMixer(const unsigned char* dataBuffer, int numBytes) : Assignme
}
void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuffer* bufferToAdd,
AvatarAudioRingBuffer* listeningNodeBuffer) {
float bearingRelativeAngleToSource = 0.0f;
float attenuationCoefficient = 1.0f;
int numSamplesDelay = 0;
float weakChannelAmplitudeRatio = 1.0f;
const int PHASE_DELAY_AT_90 = 20;
static stk::StkFrames stkFrameBuffer(BUFFER_LENGTH_SAMPLES_PER_CHANNEL, 1);
stk::TwoPole* otherNodeTwoPole = NULL;
if (bufferToAdd != listeningNodeBuffer) {
// if the two buffer pointers do no match then these are different buffers
glm::vec3 listenerPosition = listeningNodeBuffer->getPosition();
glm::vec3 relativePosition = bufferToAdd->getPosition() - listeningNodeBuffer->getPosition();
glm::quat inverseOrientation = glm::inverse(listeningNodeBuffer->getOrientation());
float distanceSquareToSource = glm::dot(relativePosition, relativePosition);
float radius = 0.0f;
if (bufferToAdd->getType() == PositionalAudioRingBuffer::Injector) {
InjectedAudioRingBuffer* injectedBuffer = (InjectedAudioRingBuffer*) bufferToAdd;
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) {
// 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);
} else {
// calculate the angle delivery for off-axis attenuation
glm::vec3 rotatedListenerPosition = glm::inverse(bufferToAdd->getOrientation()) * relativePosition;
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;
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;
}
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 = listeningNodeBuffer->getTwoPoles();
TwoPoleNodeMap::iterator twoPoleIterator = nodeTwoPoles.find(bufferToAdd);
if (twoPoleIterator == nodeTwoPoles.end()) {
// setup the freeVerb effect for this source for this client
otherNodeTwoPole = nodeTwoPoles[bufferToAdd] = 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 = bufferToAdd->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 = bufferToAdd->getNextOutput() == bufferToAdd->getBuffer()
? bufferToAdd->getBuffer() + RING_BUFFER_LENGTH_SAMPLES - numSamplesDelay
: bufferToAdd->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;
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 >= 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
bufferToAdd->getNextOutput()[s] = (int16_t) stkFrameBuffer[s];
}
}
}
void AudioMixer::prepareMixForListeningNode(Node* node) {
NodeList* nodeList = NodeList::getInstance();
AvatarAudioRingBuffer* nodeRingBuffer = ((AudioMixerClientData*) node->getLinkedData())->getAvatarAudioRingBuffer();
// 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 (otherNode->getLinkedData()) {
AudioMixerClientData* otherNodeClientData = (AudioMixerClientData*) otherNode->getLinkedData();
// enumerate the ARBs attached to the otherNode and add all that should be added to mix
for (int i = 0; i < otherNodeClientData->getRingBuffers().size(); i++) {
PositionalAudioRingBuffer* otherNodeBuffer = otherNodeClientData->getRingBuffers()[i];
if ((*otherNode != *node
|| otherNodeBuffer->getType() != PositionalAudioRingBuffer::Microphone
|| nodeRingBuffer->shouldLoopbackForNode())
&& otherNodeBuffer->willBeAddedToMix()) {
addBufferToMixForListeningNodeWithBuffer(otherNodeBuffer, nodeRingBuffer);
}
}
}
}
}
void AudioMixer::run() {
// change the logging target name while this is running
Logging::setTargetName(AUDIO_MIXER_LOGGING_TARGET_NAME);
@ -86,7 +279,7 @@ void AudioMixer::run() {
nodeList->startSilentNodeRemovalThread();
unsigned char* packetData = new unsigned char[MAX_PACKET_SIZE];
unsigned char packetData[MAX_PACKET_SIZE] = {};
sockaddr* nodeAddress = new sockaddr;
@ -100,8 +293,6 @@ void AudioMixer::run() {
unsigned char clientPacket[BUFFER_LENGTH_BYTES_STEREO + numBytesPacketHeader];
populateTypeAndVersion(clientPacket, PACKET_TYPE_MIXED_AUDIO);
int16_t clientSamples[BUFFER_LENGTH_SAMPLES_PER_CHANNEL * 2] = {};
gettimeofday(&startTime, NULL);
timeval lastDomainServerCheckIn = {};
@ -110,8 +301,6 @@ void AudioMixer::run() {
float sumFrameTimePercentages = 0.0f;
int numStatCollections = 0;
stk::StkFrames stkFrameBuffer(BUFFER_LENGTH_SAMPLES_PER_CHANNEL, 1);
// if we'll be sending stats, call the Logstash::socket() method to make it load the logstash IP outside the loop
if (Logging::shouldSendStats()) {
Logging::socket();
@ -147,209 +336,25 @@ void AudioMixer::run() {
nodeList->possiblyPingInactiveNodes();
for (NodeList::iterator node = nodeList->begin(); node != nodeList->end(); node++) {
PositionalAudioRingBuffer* positionalRingBuffer = (PositionalAudioRingBuffer*) node->getLinkedData();
if (positionalRingBuffer && positionalRingBuffer->shouldBeAddedToMix(JITTER_BUFFER_SAMPLES)) {
// this is a ring buffer that is ready to go
// set its flag so we know to push its buffer when all is said and done
positionalRingBuffer->setWillBeAddedToMix(true);
if (node->getLinkedData()) {
((AudioMixerClientData*) node->getLinkedData())->checkBuffersBeforeFrameSend(JITTER_BUFFER_SAMPLES);
}
}
for (NodeList::iterator node = nodeList->begin(); node != nodeList->end(); node++) {
const int PHASE_DELAY_AT_90 = 20;
if (node->getType() == NODE_TYPE_AGENT && node->getActiveSocket() && node->getLinkedData()) {
AvatarAudioRingBuffer* nodeRingBuffer = (AvatarAudioRingBuffer*) node->getLinkedData();
if (node->getType() == NODE_TYPE_AGENT && node->getActiveSocket() && node->getLinkedData()
&& ((AudioMixerClientData*) node->getLinkedData())->getAvatarAudioRingBuffer()) {
prepareMixForListeningNode(&(*node));
// 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 (otherNode->getLinkedData()
&& ((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...
float bearingRelativeAngleToSource = 0.0f;
float attenuationCoefficient = 1.0f;
int numSamplesDelay = 0;
float weakChannelAmplitudeRatio = 1.0f;
stk::TwoPole* otherNodeTwoPole = NULL;
if (otherNode != node) {
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;
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) {
// 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);
} 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));
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));
// 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->getUUID());
if (twoPoleIterator == nodeTwoPoles.end()) {
// setup the freeVerb effect for this source for this client
otherNodeTwoPole = nodeTwoPoles[otherNode->getUUID()] = 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;
delayedChannel[s] = glm::clamp(delayedChannel[s] + earlierSample,
MIN_SAMPLE_VALUE,
MAX_SAMPLE_VALUE);
}
int16_t currentSample = stkFrameBuffer[s] * attenuationCoefficient;
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 >= 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];
}
}
}
}
memcpy(clientPacket + numBytesPacketHeader, clientSamples, sizeof(clientSamples));
memcpy(clientPacket + numBytesPacketHeader, _clientSamples, sizeof(_clientSamples));
nodeList->getNodeSocket()->send(node->getActiveSocket(), clientPacket, sizeof(clientPacket));
}
}
// push forward the next output pointers for any audio buffers we used
for (NodeList::iterator node = nodeList->begin(); node != nodeList->end(); node++) {
PositionalAudioRingBuffer* nodeBuffer = (PositionalAudioRingBuffer*) node->getLinkedData();
if (nodeBuffer && nodeBuffer->willBeAddedToMix()) {
nodeBuffer->setNextOutput(nodeBuffer->getNextOutput() + BUFFER_LENGTH_SAMPLES_PER_CHANNEL);
if (nodeBuffer->getNextOutput() >= nodeBuffer->getBuffer() + RING_BUFFER_LENGTH_SAMPLES) {
nodeBuffer->setNextOutput(nodeBuffer->getBuffer());
}
nodeBuffer->setWillBeAddedToMix(false);
if (node->getLinkedData()) {
((AudioMixerClientData*) node->getLinkedData())->pushBuffersAfterFrameSend();
}
}
@ -367,12 +372,6 @@ void AudioMixer::run() {
if (matchingNode) {
nodeList->updateNodeWithData(matchingNode, nodeAddress, packetData, receivedBytes);
if (packetData[0] != PACKET_TYPE_INJECT_AUDIO
&& std::isnan(((PositionalAudioRingBuffer *)matchingNode->getLinkedData())->getOrientation().x)) {
// kill off this node - temporary solution to mixer crash on mac sleep
matchingNode->setAlive(false);
}
}
} else {
// let processNodeData handle it.

14
assignment-client/src/audio/AudioMixer.h
View file

@ -10,6 +10,10 @@
#define __hifi__AudioMixer__
#include <Assignment.h>
#include <AudioRingBuffer.h>
class PositionalAudioRingBuffer;
class AvatarAudioRingBuffer;
/// Handles assignments of type AudioMixer - mixing streams of audio and re-distributing to various clients.
class AudioMixer : public Assignment {
@ -18,6 +22,16 @@ public:
/// runs the audio mixer
void run();
private:
/// adds one buffer to the mix for a listening node
void addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuffer* bufferToAdd,
AvatarAudioRingBuffer* listeningNodeBuffer);
/// prepares and sends a mix to one Node
void prepareMixForListeningNode(Node* node);
int16_t _clientSamples[BUFFER_LENGTH_SAMPLES_PER_CHANNEL * 2];
};
#endif /* defined(__hifi__AudioMixer__) */

27
assignment-client/src/audio/AudioMixerClientData.cpp
View file

@ -74,3 +74,30 @@ int AudioMixerClientData::parseData(unsigned char* packetData, int numBytes) {
return 0;
}
void AudioMixerClientData::checkBuffersBeforeFrameSend(int jitterBufferLengthSamples) {
for (int i = 0; i < _ringBuffers.size(); i++) {
if (_ringBuffers[i]->shouldBeAddedToMix(jitterBufferLengthSamples)) {
// this is a ring buffer that is ready to go
// set its flag so we know to push its buffer when all is said and done
_ringBuffers[i]->setWillBeAddedToMix(true);
}
}
}
void AudioMixerClientData::pushBuffersAfterFrameSend() {
for (int i = 0; i < _ringBuffers.size(); i++) {
if (_ringBuffers[i]->willBeAddedToMix()) {
// this was a used buffer, push the output pointer forwards
PositionalAudioRingBuffer* audioBuffer = _ringBuffers[i];
audioBuffer->setNextOutput(audioBuffer->getNextOutput() + BUFFER_LENGTH_SAMPLES_PER_CHANNEL);
if (audioBuffer->getNextOutput() >= audioBuffer->getBuffer() + RING_BUFFER_LENGTH_SAMPLES) {
audioBuffer->setNextOutput(audioBuffer->getBuffer());
}
audioBuffer->setWillBeAddedToMix(false);
}
}
}

7
assignment-client/src/audio/AudioMixerClientData.h
View file

@ -20,9 +20,12 @@ class AudioMixerClientData : public NodeData {
public:
~AudioMixerClientData();
int parseData(unsigned char* packetData, int numBytes);
const std::vector<PositionalAudioRingBuffer*> getRingBuffers() const { return _ringBuffers; }
AvatarAudioRingBuffer* getAvatarAudioRingBuffer() const;
int parseData(unsigned char* packetData, int numBytes);
void checkBuffersBeforeFrameSend(int jitterBufferLengthSamples);
void pushBuffersAfterFrameSend();
private:
std::vector<PositionalAudioRingBuffer*> _ringBuffers;
};

2
assignment-client/src/audio/AvatarAudioRingBuffer.h
View file

@ -16,7 +16,7 @@
#include "PositionalAudioRingBuffer.h"
typedef std::map<QUuid, stk::TwoPole*> TwoPoleNodeMap;
typedef std::map<PositionalAudioRingBuffer*, stk::TwoPole*> TwoPoleNodeMap;
class AvatarAudioRingBuffer : public PositionalAudioRingBuffer {
public:

4
libraries/shared/src/Node.cpp
View file

@ -122,10 +122,6 @@ void Node::activatePublicSocket() {
_activeSocket = _publicSocket;
}
bool Node::operator==(const Node& otherNode) {
return matches(otherNode._publicSocket, otherNode._localSocket, otherNode._type);
}
bool Node::matches(sockaddr* otherPublicSocket, sockaddr* otherLocalSocket, char otherNodeType) {
// checks if two node objects are the same node (same type + local + public address)
return _type == otherNodeType

4
libraries/shared/src/Node.h
View file

@ -29,7 +29,8 @@ public:
Node(const QUuid& uuid, char type, sockaddr* publicSocket, sockaddr* localSocket);
~Node();
bool operator==(const Node& otherNode);
bool operator==(const Node& otherNode) const { return _uuid == otherNode._uuid; }
bool operator!=(const Node& otherNode) const { return !(*this == otherNode); }
bool matches(sockaddr* otherPublicSocket, sockaddr* otherLocalSocket, char otherNodeType);
@ -73,6 +74,7 @@ public:
void unlock() { pthread_mutex_unlock(&_mutex); }
static void printLog(Node const&);
private:
// privatize copy and assignment operator to disallow Node copying
Node(const Node &otherNode);