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remove dependency on Stk library and TwoPole effect

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This commit is contained in:
Stephen Birarda 2013-10-21 16:30:01 -07:00
parent 00446076b5
commit 73a42fbfde
3 changed files with 7 additions and 70 deletions
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60
assignment-client/src/audio/AudioMixer.cpp
View file

@ -77,10 +77,6 @@ void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuf
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 not match then these are different buffers
@ -156,26 +152,6 @@ void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuf
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);
}
}
@ -192,44 +168,22 @@ void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuf
? 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);
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);
// pull the current sample for the good channel
int16_t currentSample = sourceBuffer[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];
// place the curernt sample at the right spot in the delayed channel
int sumSample = delayedChannel[s + numSamplesDelay] + (currentSample * weakChannelAmplitudeRatio);
delayedChannel[s + numSamplesDelay] = glm::clamp(sumSample, MIN_SAMPLE_VALUE, MAX_SAMPLE_VALUE);
}
}
}

8
assignment-client/src/audio/AvatarAudioRingBuffer.cpp
View file

@ -12,18 +12,10 @@
AvatarAudioRingBuffer::AvatarAudioRingBuffer() :
PositionalAudioRingBuffer(PositionalAudioRingBuffer::Microphone),
_twoPoles(),
_shouldLoopbackForNode(false) {
}
AvatarAudioRingBuffer::~AvatarAudioRingBuffer() {
// enumerate the freeVerbs map and delete the FreeVerb objects
for (TwoPoleNodeMap::iterator poleIterator = _twoPoles.begin(); poleIterator != _twoPoles.end(); poleIterator++) {
delete poleIterator->second;
}
}
int AvatarAudioRingBuffer::parseData(unsigned char* sourceBuffer, int numBytes) {
_shouldLoopbackForNode = (sourceBuffer[0] == PACKET_TYPE_MICROPHONE_AUDIO_WITH_ECHO);
return PositionalAudioRingBuffer::parseData(sourceBuffer, numBytes);

9
assignment-client/src/audio/AvatarAudioRingBuffer.h
View file

@ -9,31 +9,22 @@
#ifndef __hifi__AvatarAudioRingBuffer__
#define __hifi__AvatarAudioRingBuffer__
#include <Stk.h>
#include <TwoPole.h>
#include <QtCore/QUuid>
#include "PositionalAudioRingBuffer.h"
typedef std::map<PositionalAudioRingBuffer*, stk::TwoPole*> TwoPoleNodeMap;
class AvatarAudioRingBuffer : public PositionalAudioRingBuffer {
public:
AvatarAudioRingBuffer();
~AvatarAudioRingBuffer();
int parseData(unsigned char* sourceBuffer, int numBytes);
TwoPoleNodeMap& getTwoPoles() { return _twoPoles; }
bool shouldLoopbackForNode() const { return _shouldLoopbackForNode; }
private:
// disallow copying of AvatarAudioRingBuffer objects
AvatarAudioRingBuffer(const AvatarAudioRingBuffer&);
AvatarAudioRingBuffer& operator= (const AvatarAudioRingBuffer&);
TwoPoleNodeMap _twoPoles;
bool _shouldLoopbackForNode;
};