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remove the CPU hogging FreeVerb effect

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This commit is contained in:
Stephen Birarda 2013-06-06 11:45:33 -07:00
parent b444933810
commit 19ab1816bb
3 changed files with 0 additions and 68 deletions
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8
audio-mixer/src/AvatarAudioRingBuffer.cpp
View file

@ -11,18 +11,10 @@
#include "AvatarAudioRingBuffer.h" #include "AvatarAudioRingBuffer.h"
AvatarAudioRingBuffer::AvatarAudioRingBuffer() : AvatarAudioRingBuffer::AvatarAudioRingBuffer() :
_freeVerbs(),
_shouldLoopbackForAgent(false) { _shouldLoopbackForAgent(false) {
} }
AvatarAudioRingBuffer::~AvatarAudioRingBuffer() {
// enumerate the freeVerbs map and delete the FreeVerb objects
for (FreeVerbAgentMap::iterator verbIterator = _freeVerbs.begin(); verbIterator != _freeVerbs.end(); verbIterator++) {
delete verbIterator->second;
}
}
int AvatarAudioRingBuffer::parseData(unsigned char* sourceBuffer, int numBytes) { int AvatarAudioRingBuffer::parseData(unsigned char* sourceBuffer, int numBytes) {
_shouldLoopbackForAgent = (sourceBuffer[0] == PACKET_HEADER_MICROPHONE_AUDIO_WITH_ECHO); _shouldLoopbackForAgent = (sourceBuffer[0] == PACKET_HEADER_MICROPHONE_AUDIO_WITH_ECHO);
return PositionalAudioRingBuffer::parseData(sourceBuffer, numBytes); return PositionalAudioRingBuffer::parseData(sourceBuffer, numBytes);

8
audio-mixer/src/AvatarAudioRingBuffer.h
View file

@ -9,28 +9,20 @@
#ifndef __hifi__AvatarAudioRingBuffer__ #ifndef __hifi__AvatarAudioRingBuffer__
#define __hifi__AvatarAudioRingBuffer__ #define __hifi__AvatarAudioRingBuffer__
#include <Stk.h>
#include <FreeVerb.h>
#include "PositionalAudioRingBuffer.h" #include "PositionalAudioRingBuffer.h"
typedef std::map<uint16_t, stk::FreeVerb*> FreeVerbAgentMap;
class AvatarAudioRingBuffer : public PositionalAudioRingBuffer { class AvatarAudioRingBuffer : public PositionalAudioRingBuffer {
public: public:
AvatarAudioRingBuffer(); AvatarAudioRingBuffer();
~AvatarAudioRingBuffer();
int parseData(unsigned char* sourceBuffer, int numBytes); int parseData(unsigned char* sourceBuffer, int numBytes);
FreeVerbAgentMap& getFreeVerbs() { return _freeVerbs; }
bool shouldLoopbackForAgent() const { return _shouldLoopbackForAgent; } bool shouldLoopbackForAgent() const { return _shouldLoopbackForAgent; }
private: private:
// disallow copying of AvatarAudioRingBuffer objects // disallow copying of AvatarAudioRingBuffer objects
AvatarAudioRingBuffer(const AvatarAudioRingBuffer&); AvatarAudioRingBuffer(const AvatarAudioRingBuffer&);
AvatarAudioRingBuffer& operator= (const AvatarAudioRingBuffer&); AvatarAudioRingBuffer& operator= (const AvatarAudioRingBuffer&);
FreeVerbAgentMap _freeVerbs;
bool _shouldLoopbackForAgent; bool _shouldLoopbackForAgent;
}; };

52
audio-mixer/src/main.cpp
View file

@ -104,11 +104,6 @@ int main(int argc, const char* argv[]) {
int16_t clientSamples[BUFFER_LENGTH_SAMPLES_PER_CHANNEL * 2] = {}; int16_t clientSamples[BUFFER_LENGTH_SAMPLES_PER_CHANNEL * 2] = {};
// setup STK for the reverb effect
const float DISTANCE_REVERB_DAMPING = 0.6f;
const float DISTANCE_REVERB_ROOM_SIZE = 0.75f;
const float DISTANCE_REVERB_WIDTH = 0.5f;
gettimeofday(&startTime, NULL); gettimeofday(&startTime, NULL);
while (true) { while (true) {
@ -142,8 +137,6 @@ int main(int argc, const char* argv[]) {
int numSamplesDelay = 0; int numSamplesDelay = 0;
float weakChannelAmplitudeRatio = 1.0f; float weakChannelAmplitudeRatio = 1.0f;
stk::FreeVerb* otherAgentFreeVerb = NULL;
if (otherAgent != agent) { if (otherAgent != agent) {
glm::vec3 listenerPosition = agentRingBuffer->getPosition(); glm::vec3 listenerPosition = agentRingBuffer->getPosition();
@ -220,40 +213,6 @@ int main(int argc, const char* argv[]) {
numSamplesDelay = PHASE_DELAY_AT_90 * sinRatio; numSamplesDelay = PHASE_DELAY_AT_90 * sinRatio;
weakChannelAmplitudeRatio = 1 - (PHASE_AMPLITUDE_RATIO_AT_90 * sinRatio); weakChannelAmplitudeRatio = 1 - (PHASE_AMPLITUDE_RATIO_AT_90 * sinRatio);
} }
//
// FreeVerbAgentMap& agentFreeVerbs = agentRingBuffer->getFreeVerbs();
// FreeVerbAgentMap::iterator freeVerbIterator = agentFreeVerbs.find(otherAgent->getAgentID());
//
// if (freeVerbIterator == agentFreeVerbs.end()) {
// // setup the freeVerb effect for this source for this client
// otherAgentFreeVerb = agentFreeVerbs[otherAgent->getAgentID()] = new stk::FreeVerb;
//
// otherAgentFreeVerb->setDamping(DISTANCE_REVERB_DAMPING);
// otherAgentFreeVerb->setRoomSize(DISTANCE_REVERB_ROOM_SIZE);
// otherAgentFreeVerb->setWidth(DISTANCE_REVERB_WIDTH);
// } else {
// otherAgentFreeVerb = freeVerbIterator->second;
// }
//
// const float WETNESS_DOUBLING_DISTANCE_FACTOR = 2.0f;
// const float MAX_REVERB_DISTANCE = 160.0f;
//
// // higher value increases wetness more quickly with distance
// const float WETNESS_CALC_EXPONENT_BASE = 2.0f;
//
// const float MAX_EXPONENT = logf(MAX_REVERB_DISTANCE) / logf(WETNESS_DOUBLING_DISTANCE_FACTOR);
// const int MAX_EXPONENT_INT = floorf(MAX_EXPONENT);
// const float DISTANCE_REVERB_LOG_REMAINDER = fmodf(MAX_EXPONENT, MAX_EXPONENT_INT);
// const float DISTANCE_REVERB_MAX_WETNESS = 1.0f;
// const float EFFECT_MIX_RHS = DISTANCE_REVERB_MAX_WETNESS / powf(WETNESS_DOUBLING_DISTANCE_FACTOR,
// MAX_EXPONENT_INT);
//
// float effectMix = powf(WETNESS_CALC_EXPONENT_BASE,
// (0.5f * logf(distanceSquareToSource) / logf(WETNESS_CALC_EXPONENT_BASE))
// - DISTANCE_REVERB_LOG_REMAINDER);
// effectMix *= EFFECT_MIX_RHS;
//
// otherAgentFreeVerb->setEffectMix(effectMix);
} }
int16_t* goodChannel = (bearingRelativeAngleToSource > 0.0f) int16_t* goodChannel = (bearingRelativeAngleToSource > 0.0f)
@ -280,17 +239,6 @@ int main(int argc, const char* argv[]) {
int16_t currentSample = otherAgentBuffer->getNextOutput()[s]; 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; currentSample *= attenuationCoefficient;
plateauAdditionOfSamples(goodChannel[s], currentSample); plateauAdditionOfSamples(goodChannel[s], currentSample);