Mirrors/overte
3
0
Fork 0
mirror of https://github.com/overte-org/overte.git synced 2025-08-09 21:12:53 +02:00
Code Issues Projects Releases Packages Wiki Activity Actions 2

Merge pull request #497 from birarda/synthesis

Browse source 
remove the CPU hog that is FreeVerb for now
This commit is contained in:
ZappoMan 2013-06-06 11:55:57 -07:00
commit 3fda876c91
4 changed files with 3 additions and 74 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

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;
}; };

59
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) {
@ -129,8 +124,6 @@ int main(int argc, const char* argv[]) {
// zero out the client mix for this agent // zero out the client mix for this agent
memset(clientSamples, 0, sizeof(clientSamples)); memset(clientSamples, 0, sizeof(clientSamples));
const int PHASE_DELAY_AT_90 = 20;
for (AgentList::iterator otherAgent = agentList->begin(); otherAgent != agentList->end(); otherAgent++) { for (AgentList::iterator otherAgent = agentList->begin(); otherAgent != agentList->end(); otherAgent++) {
if (((PositionalAudioRingBuffer*) otherAgent->getLinkedData())->willBeAddedToMix() if (((PositionalAudioRingBuffer*) otherAgent->getLinkedData())->willBeAddedToMix()
&& (otherAgent != agent || (otherAgent == agent && agentRingBuffer->shouldLoopbackForAgent()))) { && (otherAgent != agent || (otherAgent == agent && agentRingBuffer->shouldLoopbackForAgent()))) {
@ -142,8 +135,6 @@ 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();
@ -212,6 +203,7 @@ int main(int argc, const char* argv[]) {
glm::normalize(rotatedSourcePosition), glm::normalize(rotatedSourcePosition),
glm::vec3(0.0f, 1.0f, 0.0f)); glm::vec3(0.0f, 1.0f, 0.0f));
const int PHASE_DELAY_AT_90 = 20;
const float PHASE_AMPLITUDE_RATIO_AT_90 = 0.5; const float PHASE_AMPLITUDE_RATIO_AT_90 = 0.5;
// figure out the number of samples of delay and the ratio of the amplitude // figure out the number of samples of delay and the ratio of the amplitude
@ -220,40 +212,6 @@ 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)
@ -278,20 +236,7 @@ int main(int argc, const char* argv[]) {
plateauAdditionOfSamples(delayedChannel[s], earlierSample); plateauAdditionOfSamples(delayedChannel[s], earlierSample);
} }
int16_t currentSample = otherAgentBuffer->getNextOutput()[s]; int16_t currentSample = otherAgentBuffer->getNextOutput()[s] * attenuationCoefficient;
// 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); plateauAdditionOfSamples(goodChannel[s], currentSample);

2
interface/src/Audio.cpp
View file

@ -129,7 +129,7 @@ int audioCallback (const void* inputBuffer,
glm::vec3 headPosition = interfaceAvatar->getHeadJointPosition(); glm::vec3 headPosition = interfaceAvatar->getHeadJointPosition();
glm::quat headOrientation = interfaceAvatar->getHead().getOrientation(); glm::quat headOrientation = interfaceAvatar->getHead().getOrientation();
int leadingBytes = 1 + sizeof(headPosition) + sizeof(headOrientation); int leadingBytes = sizeof(PACKET_HEADER_MICROPHONE_AUDIO_NO_ECHO) + sizeof(headPosition) + sizeof(headOrientation);
// we need the amount of bytes in the buffer + 1 for type // we need the amount of bytes in the buffer + 1 for type
// + 12 for 3 floats for position + float for bearing + 1 attenuation byte // + 12 for 3 floats for position + float for bearing + 1 attenuation byte