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refactoring in the audio-mixer for spherical injector and FreeVerb

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This commit is contained in:
Stephen Birarda 2013-06-05 15:55:39 -07:00
parent 2db7b5de4e
commit 60b018b8b2
1 changed files with 19 additions and 21 deletions
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40
audio-mixer/src/main.cpp
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@ -129,6 +129,8 @@ int main(int argc, const char* argv[]) {
// zero out the client mix for this agent
memset(clientSamples, 0, sizeof(clientSamples));
const int PHASE_DELAY_AT_90 = 20;
for (AgentList::iterator otherAgent = agentList->begin(); otherAgent != agentList->end(); otherAgent++) {
if (((PositionalAudioRingBuffer*) otherAgent->getLinkedData())->willBeAddedToMix()
&& (otherAgent != agent || (otherAgent == agent && agentRingBuffer->shouldLoopbackForAgent()))) {
@ -143,10 +145,10 @@ int main(int argc, const char* argv[]) {
stk::FreeVerb* otherAgentFreeVerb = NULL;
if (otherAgent != agent) {
glm::vec3 listenerPosition = agentRingBuffer->getPosition();
glm::vec3 relativePosition = otherAgentBuffer->getPosition() - agentRingBuffer->getPosition();
glm::quat inverseOrientation = glm::inverse(agentRingBuffer->getOrientation());
glm::vec3 rotatedSourcePosition = inverseOrientation * relativePosition;
float distanceSquareToSource = glm::dot(relativePosition, relativePosition);
float radius = 0.0f;
@ -161,19 +163,10 @@ int main(int argc, const char* argv[]) {
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
// multiply the normalized vector between the center of the sphere
// and the position of the source by the radius to get the
// closest point on the boundary of the sphere to the source
glm::vec3 closestPoint = glm::normalize(relativePosition) * radius;
// for the other calculations the agent position is the closest point on the sphere
rotatedSourcePosition = inverseOrientation * closestPoint;
// ovveride the distance to the agent with the distance to the point on the
// boundary of the sphere
distanceSquareToSource = glm::distance2(listenerPosition, -closestPoint);
distanceSquareToSource -= (radius * radius);
} else {
// calculate the angle delivery for off-axis attenuation
@ -193,6 +186,8 @@ int main(int argc, const char* argv[]) {
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;
@ -216,7 +211,6 @@ int main(int argc, const char* argv[]) {
glm::vec3(0.0f, 1.0f, 0.0f));
const float PHASE_AMPLITUDE_RATIO_AT_90 = 0.5;
const int PHASE_DELAY_AT_90 = 20;
// figure out the number of samples of delay and the ratio of the amplitude
// in the weak channel for audio spatialization
@ -245,7 +239,7 @@ int main(int argc, const char* argv[]) {
float effectMix = powf(2.0f, (0.5f * logf(distanceSquareToSource)
/ logf(2.0f)) - DISTANCE_REVERB_LOG_REMAINDER)
* DISTANCE_REVERB_MAX_WETNESS / 64.0f;;
* DISTANCE_REVERB_MAX_WETNESS / 64.0f;
otherAgentFreeVerb->setEffectMix(effectMix);
}
@ -266,24 +260,28 @@ int main(int argc, const char* argv[]) {
if (s < numSamplesDelay) {
// pull the earlier sample for the delayed channel
int earlierSample = delaySamplePointer[s]
* attenuationCoefficient
* weakChannelAmplitudeRatio;
* attenuationCoefficient
* weakChannelAmplitudeRatio;
// apply the STK FreeVerb effect
if (otherAgentFreeVerb) {
earlierSample = otherAgentFreeVerb->tick(earlierSample);
}
plateauAdditionOfSamples(delayedChannel[s], earlierSample);
}
int16_t currentSample = (otherAgentBuffer->getNextOutput()[s] * attenuationCoefficient);
int16_t currentSample = otherAgentBuffer->getNextOutput()[s];
// apply the STK FreeVerb effect
if (otherAgentFreeVerb) {
currentSample = otherAgentFreeVerb->tick(currentSample);
if (s >= BUFFER_LENGTH_SAMPLES_PER_CHANNEL - PHASE_DELAY_AT_90) {
// there is the possiblity this will be re-used as a delayed sample
// so store the reverbed sample so that is what will be pulled
otherAgentBuffer->getNextOutput()[s] = currentSample;
}
}
currentSample *= attenuationCoefficient;
plateauAdditionOfSamples(goodChannel[s], currentSample);
if (s + numSamplesDelay < BUFFER_LENGTH_SAMPLES_PER_CHANNEL) {