Merge pull request #3423 from chansensturm/audio-temp

Head Penumbra filter changes (feedback from testing with Freddy)

assignment-client/src/audio/AudioMixer.cpp

@@ -321,9 +321,14 @@ int AudioMixer::addStreamToMixForListeningNodeWithStream(PositionalAudioStream*
         const float TWO_OVER_PI = 2.0f / PI;
         
         const float ZERO_DB = 1.0f;
-        const float NEGATIVE_ONE_DB = 0.891f;
+//        const float NEGATIVE_ONE_DB = 0.891f;
         const float NEGATIVE_THREE_DB = 0.708f;
-
+        const float NEGATIVE_SIX_DB = 0.501f;
+        
+        const float FILTER_GAIN_AT_0 = ZERO_DB; // source is in front
+        const float FILTER_GAIN_AT_90 = NEGATIVE_SIX_DB; // source is incident to left or right ear
+        const float FILTER_GAIN_AT_180 = NEGATIVE_SIX_DB; // source is behind
+        
         const float FILTER_CUTOFF_FREQUENCY_HZ = 1000.0f;
         
         const float penumbraFilterFrequency = FILTER_CUTOFF_FREQUENCY_HZ; // constant frequency
@@ -334,32 +339,25 @@ int AudioMixer::addStreamToMixForListeningNodeWithStream(PositionalAudioStream*
 
         // variable gain calculation broken down by quadrent
         if (bearingAngleToSource < -PI_OVER_TWO && bearingAngleToSource > -PI) {
-            // gainL(-pi/2,0b)->(-pi,-1db)
             penumbraFilterGainL = TWO_OVER_PI * 
-                (ZERO_DB - NEGATIVE_ONE_DB) * (bearingAngleToSource + PI_OVER_TWO) + ZERO_DB;
-            // gainR(-pi/2,-3db)->(-pi,-1db)
+                (FILTER_GAIN_AT_0 - FILTER_GAIN_AT_180) * (bearingAngleToSource + PI_OVER_TWO) + FILTER_GAIN_AT_0;
             penumbraFilterGainR = TWO_OVER_PI * 
-                (NEGATIVE_THREE_DB - NEGATIVE_ONE_DB) * (bearingAngleToSource + PI_OVER_TWO) + NEGATIVE_THREE_DB;
+                (FILTER_GAIN_AT_90 - FILTER_GAIN_AT_180) * (bearingAngleToSource + PI_OVER_TWO) + FILTER_GAIN_AT_90;
         } else if (bearingAngleToSource <= PI && bearingAngleToSource > PI_OVER_TWO) {
-            // gainL(+pi,-1db)->(pi/2,-3db)
             penumbraFilterGainL = TWO_OVER_PI * 
-                (NEGATIVE_ONE_DB - NEGATIVE_THREE_DB) * (bearingAngleToSource - PI) + NEGATIVE_ONE_DB;
-            // gainR(+pi,-1db)->(pi/2,0db)
+                (FILTER_GAIN_AT_180 - FILTER_GAIN_AT_90) * (bearingAngleToSource - PI) + FILTER_GAIN_AT_180;
             penumbraFilterGainR = TWO_OVER_PI * 
-                (NEGATIVE_ONE_DB - ZERO_DB) * (bearingAngleToSource - PI) + NEGATIVE_ONE_DB;
+                (FILTER_GAIN_AT_180 - FILTER_GAIN_AT_0) * (bearingAngleToSource - PI) + FILTER_GAIN_AT_180;
         } else if (bearingAngleToSource <= PI_OVER_TWO && bearingAngleToSource > 0) {
-            // gainL(+pi/2,-3db)->(0,0db)
             penumbraFilterGainL = TWO_OVER_PI *
-                (NEGATIVE_THREE_DB - ZERO_DB) * (bearingAngleToSource - PI_OVER_TWO) + NEGATIVE_THREE_DB;
-            // gainR(+p1/2,0db)->(0,0db)
-            penumbraFilterGainR = ZERO_DB;            
+                (FILTER_GAIN_AT_90 - FILTER_GAIN_AT_0) * (bearingAngleToSource - PI_OVER_TWO) + FILTER_GAIN_AT_90;
+            penumbraFilterGainR = FILTER_GAIN_AT_0;            
         } else {
-            // gainL(0,0db)->(-pi/2,0db)
-            penumbraFilterGainL = ZERO_DB;
-            // gainR(0,0db)->(-pi/2,-3db)
+            penumbraFilterGainL = FILTER_GAIN_AT_0;
             penumbraFilterGainR =  TWO_OVER_PI * 
-                (ZERO_DB - NEGATIVE_THREE_DB) * (bearingAngleToSource) + ZERO_DB;
+                (FILTER_GAIN_AT_0 - FILTER_GAIN_AT_90) * (bearingAngleToSource) + FILTER_GAIN_AT_0;
         }
+
 #if 0
             qDebug() << "avatar="
                      << listeningNodeStream

domain-server/resources/web/settings/describe.json

@@ -67,7 +67,7 @@
 	    "type": "checkbox",
         "label": "Enable Positional Filter",
         "help": "If enabled, positional audio stream uses lowpass filter",
-        "default": false
+        "default": true
       }
     }
   }

libraries/audio/src/AudioSourceTone.cpp

@@ -0,0 +1,59 @@
+//
+//  AudioSourceTone.cpp
+//  hifi
+//
+//  Created by Craig Hansen-Sturm on 8/10/14.
+//  Copyright 2014 High Fidelity, Inc.
+//
+//  Distributed under the Apache License, Version 2.0.
+//  See the accompanying file LICENSE or http://www.apache.org/licenses/LICENSE-2.0.html
+//
+
+#include <assert.h>
+#include <math.h>
+#include <SharedUtil.h>
+#include "AudioRingBuffer.h"
+#include "AudioFormat.h"
+#include "AudioBuffer.h"
+#include "AudioSourceTone.h"
+
+AudioSourceTone::AudioSourceTone() {
+    initialize();
+}
+
+AudioSourceTone::~AudioSourceTone() {
+    finalize();
+}
+
+void AudioSourceTone::finalize() {
+}
+
+void AudioSourceTone::reset() {
+}
+
+void AudioSourceTone::updateCoefficients() {
+    _omega = _frequency / _sampleRate * TWO_PI;
+    _epsilon = 2.0f * sinf(_omega / 2.0f);
+    _yq1 = cosf(-1.0f * _omega);
+    _y1 = sinf(+1.0f * _omega);   
+}
+
+void AudioSourceTone::initialize() {
+    const float32_t FREQUENCY_220_HZ = 220.0f;
+    const float32_t GAIN_MINUS_3DB = 0.708f;
+    setParameters(SAMPLE_RATE, FREQUENCY_220_HZ, GAIN_MINUS_3DB);
+}
+
+void AudioSourceTone::setParameters(const float32_t sampleRate, const float32_t frequency,  const float32_t amplitude) {
+    _sampleRate = std::max(sampleRate, 1.0f);
+    _frequency = std::max(frequency, 1.0f);
+    _amplitude = std::max(amplitude, 1.0f);
+    updateCoefficients();
+}
+
+void AudioSourceTone::getParameters(float32_t& sampleRate, float32_t& frequency, float32_t& amplitude) {
+    sampleRate = _sampleRate;
+    frequency = _frequency;
+    amplitude = _amplitude;
+}
+

libraries/audio/src/AudioSourceTone.h

@@ -23,67 +23,41 @@ class AudioSourceTone
     float32_t _yq1;  
     float32_t _y1;
     
-    void updateCoefficients() {
-        _omega = _frequency / _sampleRate * TWO_PI;
-        _epsilon = 2.0f * sinf(_omega / 2.0f);
-        _yq1 = cosf(-1.0f * _omega);
-        _y1 = sinf(+1.0f * _omega);
-    }
+    void updateCoefficients();
     
 public:
-    AudioSourceTone() {
-        initialize();
-    }
+    AudioSourceTone();
+    ~AudioSourceTone();
     
-    ~AudioSourceTone() {
-        finalize();
-    }
-    
-    void initialize() {
-        const float32_t FREQUENCY_220_HZ = 220.0f;
-        const float32_t GAIN_MINUS_3DB = 0.708f;
-        setParameters(SAMPLE_RATE, FREQUENCY_220_HZ, GAIN_MINUS_3DB);
-    }
-    
-    void finalize() {
-    }
-    
-    void reset() {
-    }
+    void initialize();
+    void finalize();
+    void reset();
 
-    void setParameters(const float32_t sampleRate, const float32_t frequency,  const float32_t amplitude) {
-        _sampleRate = std::max(sampleRate, 1.0f);
-        _frequency = std::max(frequency, 1.0f);
-        _amplitude = std::max(amplitude, 1.0f);
-        updateCoefficients();
-    }
+    void setParameters(const float32_t sampleRate, const float32_t frequency,  const float32_t amplitude);    
+    void getParameters(float32_t& sampleRate, float32_t& frequency, float32_t& amplitude);
     
-    void getParameters(float32_t& sampleRate, float32_t& frequency, float32_t& amplitude) {
-        sampleRate = _sampleRate;
-        frequency = _frequency;
-        amplitude = _amplitude;
-    }
+    void render(AudioBufferFloat32& frameBuffer);
+};
 
-    void render(AudioBufferFloat32& frameBuffer) {
+
+inline void AudioSourceTone::render(AudioBufferFloat32& frameBuffer) {
+    float32_t** samples = frameBuffer.getFrameData();
+    float32_t yq;
+    float32_t y;
+    for (uint16_t i = 0; i < frameBuffer.getFrameCount(); ++i) {
         
-        float32_t** samples = frameBuffer.getFrameData();
-        float32_t yq;
-        float32_t y;
-        for (uint16_t i = 0; i < frameBuffer.getFrameCount(); ++i) {
-            
-            yq = _yq1 - (_epsilon * _y1);
-            y = _y1 + (_epsilon * yq);
-            
-            // update delays
-            _yq1 = yq;
-            _y1 = y;
-            
-            for (uint16_t j = 0; j < frameBuffer.getChannelCount(); ++j) {
-                samples[j][i] = _amplitude * y;
-            }
+        yq = _yq1 - (_epsilon * _y1);
+        y = _y1 + (_epsilon * yq);
+        
+        // update delays
+        _yq1 = yq;
+        _y1 = y;
+        
+        for (uint16_t j = 0; j < frameBuffer.getChannelCount(); ++j) {
+            samples[j][i] = _amplitude * y;
         }
     }
-};
+}
 
 #endif