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https://github.com/overte-org/overte.git
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Merge branch 'master' of https://github.com/worklist/hifi into gyroquat
This commit is contained in:
Andrzej Kapolka
commit
7f8b6fe0a6
8 changed files with 6 additions and 103 deletions
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@ -18,10 +18,4 @@ include_glm(${TARGET_NAME} ${ROOT_DIR})
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# link the shared hifi library
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include(${MACRO_DIR}/LinkHifiLibrary.cmake)
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link_hifi_library(shared ${TARGET_NAME} ${ROOT_DIR})
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link_hifi_library(audio ${TARGET_NAME} ${ROOT_DIR})
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# link the stk library
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set(STK_ROOT_DIR ${ROOT_DIR}/externals/stk)
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find_package(STK REQUIRED)
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target_link_libraries(${TARGET_NAME} ${STK_LIBRARIES})
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include_directories(${STK_INCLUDE_DIRS})
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link_hifi_library(audio ${TARGET_NAME} ${ROOT_DIR})
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@ -11,18 +11,10 @@
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#include "AvatarAudioRingBuffer.h"
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AvatarAudioRingBuffer::AvatarAudioRingBuffer() :
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_freeVerbs(),
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_shouldLoopbackForAgent(false) {
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}
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AvatarAudioRingBuffer::~AvatarAudioRingBuffer() {
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// enumerate the freeVerbs map and delete the FreeVerb objects
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for (FreeVerbAgentMap::iterator verbIterator = _freeVerbs.begin(); verbIterator != _freeVerbs.end(); verbIterator++) {
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delete verbIterator->second;
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}
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}
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int AvatarAudioRingBuffer::parseData(unsigned char* sourceBuffer, int numBytes) {
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_shouldLoopbackForAgent = (sourceBuffer[0] == PACKET_HEADER_MICROPHONE_AUDIO_WITH_ECHO);
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return PositionalAudioRingBuffer::parseData(sourceBuffer, numBytes);
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@ -9,28 +9,20 @@
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#ifndef __hifi__AvatarAudioRingBuffer__
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#define __hifi__AvatarAudioRingBuffer__
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#include <Stk.h>
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#include <FreeVerb.h>
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#include "PositionalAudioRingBuffer.h"
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typedef std::map<uint16_t, stk::FreeVerb*> FreeVerbAgentMap;
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class AvatarAudioRingBuffer : public PositionalAudioRingBuffer {
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public:
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AvatarAudioRingBuffer();
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~AvatarAudioRingBuffer();
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int parseData(unsigned char* sourceBuffer, int numBytes);
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FreeVerbAgentMap& getFreeVerbs() { return _freeVerbs; }
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bool shouldLoopbackForAgent() const { return _shouldLoopbackForAgent; }
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private:
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// disallow copying of AvatarAudioRingBuffer objects
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AvatarAudioRingBuffer(const AvatarAudioRingBuffer&);
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AvatarAudioRingBuffer& operator= (const AvatarAudioRingBuffer&);
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FreeVerbAgentMap _freeVerbs;
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bool _shouldLoopbackForAgent;
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};
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@ -24,8 +24,6 @@
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#include <AgentTypes.h>
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#include <SharedUtil.h>
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#include <StdDev.h>
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#include <Stk.h>
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#include <FreeVerb.h>
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#include "InjectedAudioRingBuffer.h"
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#include "AvatarAudioRingBuffer.h"
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@ -104,11 +102,6 @@ int main(int argc, const char* argv[]) {
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int16_t clientSamples[BUFFER_LENGTH_SAMPLES_PER_CHANNEL * 2] = {};
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// setup STK for the reverb effect
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const float DISTANCE_REVERB_DAMPING = 0.6f;
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const float DISTANCE_REVERB_ROOM_SIZE = 0.75f;
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const float DISTANCE_REVERB_WIDTH = 0.5f;
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gettimeofday(&startTime, NULL);
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while (true) {
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@ -129,8 +122,6 @@ int main(int argc, const char* argv[]) {
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// zero out the client mix for this agent
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memset(clientSamples, 0, sizeof(clientSamples));
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const int PHASE_DELAY_AT_90 = 20;
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for (AgentList::iterator otherAgent = agentList->begin(); otherAgent != agentList->end(); otherAgent++) {
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if (((PositionalAudioRingBuffer*) otherAgent->getLinkedData())->willBeAddedToMix()
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&& (otherAgent != agent || (otherAgent == agent && agentRingBuffer->shouldLoopbackForAgent()))) {
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@ -142,8 +133,6 @@ int main(int argc, const char* argv[]) {
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int numSamplesDelay = 0;
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float weakChannelAmplitudeRatio = 1.0f;
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stk::FreeVerb* otherAgentFreeVerb = NULL;
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if (otherAgent != agent) {
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glm::vec3 listenerPosition = agentRingBuffer->getPosition();
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@ -212,6 +201,7 @@ int main(int argc, const char* argv[]) {
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glm::normalize(rotatedSourcePosition),
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glm::vec3(0.0f, 1.0f, 0.0f));
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const int PHASE_DELAY_AT_90 = 20;
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const float PHASE_AMPLITUDE_RATIO_AT_90 = 0.5;
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// figure out the number of samples of delay and the ratio of the amplitude
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@ -220,40 +210,6 @@ int main(int argc, const char* argv[]) {
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numSamplesDelay = PHASE_DELAY_AT_90 * sinRatio;
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weakChannelAmplitudeRatio = 1 - (PHASE_AMPLITUDE_RATIO_AT_90 * sinRatio);
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}
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FreeVerbAgentMap& agentFreeVerbs = agentRingBuffer->getFreeVerbs();
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FreeVerbAgentMap::iterator freeVerbIterator = agentFreeVerbs.find(otherAgent->getAgentID());
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if (freeVerbIterator == agentFreeVerbs.end()) {
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// setup the freeVerb effect for this source for this client
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otherAgentFreeVerb = agentFreeVerbs[otherAgent->getAgentID()] = new stk::FreeVerb;
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otherAgentFreeVerb->setDamping(DISTANCE_REVERB_DAMPING);
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otherAgentFreeVerb->setRoomSize(DISTANCE_REVERB_ROOM_SIZE);
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otherAgentFreeVerb->setWidth(DISTANCE_REVERB_WIDTH);
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} else {
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otherAgentFreeVerb = freeVerbIterator->second;
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}
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const float WETNESS_DOUBLING_DISTANCE_FACTOR = 2.0f;
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const float MAX_REVERB_DISTANCE = 160.0f;
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// higher value increases wetness more quickly with distance
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const float WETNESS_CALC_EXPONENT_BASE = 2.0f;
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const float MAX_EXPONENT = logf(MAX_REVERB_DISTANCE) / logf(WETNESS_DOUBLING_DISTANCE_FACTOR);
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const int MAX_EXPONENT_INT = floorf(MAX_EXPONENT);
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const float DISTANCE_REVERB_LOG_REMAINDER = fmodf(MAX_EXPONENT, MAX_EXPONENT_INT);
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const float DISTANCE_REVERB_MAX_WETNESS = 1.0f;
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const float EFFECT_MIX_RHS = DISTANCE_REVERB_MAX_WETNESS / powf(WETNESS_DOUBLING_DISTANCE_FACTOR,
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MAX_EXPONENT_INT);
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float effectMix = powf(WETNESS_CALC_EXPONENT_BASE,
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(0.5f * logf(distanceSquareToSource) / logf(WETNESS_CALC_EXPONENT_BASE))
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- DISTANCE_REVERB_LOG_REMAINDER);
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effectMix *= EFFECT_MIX_RHS;
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otherAgentFreeVerb->setEffectMix(effectMix);
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}
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int16_t* goodChannel = (bearingRelativeAngleToSource > 0.0f)
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@ -278,20 +234,7 @@ int main(int argc, const char* argv[]) {
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plateauAdditionOfSamples(delayedChannel[s], earlierSample);
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}
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int16_t currentSample = otherAgentBuffer->getNextOutput()[s];
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// apply the STK FreeVerb effect
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if (otherAgentFreeVerb) {
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currentSample = otherAgentFreeVerb->tick(currentSample);
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if (s >= BUFFER_LENGTH_SAMPLES_PER_CHANNEL - PHASE_DELAY_AT_90) {
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// there is the possiblity this will be re-used as a delayed sample
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// so store the reverbed sample so that is what will be pulled
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otherAgentBuffer->getNextOutput()[s] = currentSample;
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}
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}
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currentSample *= attenuationCoefficient;
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int16_t currentSample = otherAgentBuffer->getNextOutput()[s] * attenuationCoefficient;
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plateauAdditionOfSamples(goodChannel[s], currentSample);
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@ -19,10 +19,4 @@ include_glm(${TARGET_NAME} ${ROOT_DIR})
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include(${MACRO_DIR}/LinkHifiLibrary.cmake)
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link_hifi_library(shared ${TARGET_NAME} ${ROOT_DIR})
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link_hifi_library(audio ${TARGET_NAME} ${ROOT_DIR})
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link_hifi_library(avatars ${TARGET_NAME} ${ROOT_DIR})
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# link the stk library
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set(STK_ROOT_DIR ${ROOT_DIR}/externals/stk)
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find_package(STK REQUIRED)
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target_link_libraries(${TARGET_NAME} ${STK_LIBRARIES})
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include_directories(${STK_INCLUDE_DIRS})
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link_hifi_library(avatars ${TARGET_NAME} ${ROOT_DIR})
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@ -146,12 +146,6 @@ else (WIN32)
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include_directories(${PORTAUDIO_INCLUDE_DIRS})
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target_link_libraries(${TARGET_NAME} ${PORTAUDIO_LIBRARIES})
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# link the stk library
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set(STK_ROOT_DIR ${ROOT_DIR}/externals/stk)
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find_package(STK REQUIRED)
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target_link_libraries(${TARGET_NAME} ${STK_LIBRARIES})
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include_directories(${STK_INCLUDE_DIRS})
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# link required libraries on UNIX
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if (UNIX AND NOT APPLE)
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find_package(Threads REQUIRED)
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@ -129,7 +129,7 @@ int audioCallback (const void* inputBuffer,
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glm::vec3 headPosition = interfaceAvatar->getHeadJointPosition();
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glm::quat headOrientation = interfaceAvatar->getHead().getOrientation();
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int leadingBytes = 1 + sizeof(headPosition) + sizeof(headOrientation);
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int leadingBytes = sizeof(PACKET_HEADER_MICROPHONE_AUDIO_NO_ECHO) + sizeof(headPosition) + sizeof(headOrientation);
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// we need the amount of bytes in the buffer + 1 for type
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// + 12 for 3 floats for position + float for bearing + 1 attenuation byte
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@ -19,10 +19,4 @@ link_hifi_library(shared ${TARGET_NAME} ${ROOT_DIR})
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# link the threads library
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find_package(Threads REQUIRED)
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target_link_libraries(${TARGET_NAME} ${CMAKE_THREAD_LIBS_INIT})
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# link the stk library
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set(STK_ROOT_DIR ${ROOT_DIR}/externals/stk)
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find_package(STK REQUIRED)
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target_link_libraries(${TARGET_NAME} ${STK_LIBRARIES})
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include_directories(${STK_INCLUDE_DIRS})
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target_link_libraries(${TARGET_NAME} ${CMAKE_THREAD_LIBS_INIT})
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