Merge branch 'master' of https://github.com/worklist/hifi

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
       }
     }
   }

examples/loadScriptFromMessage.js

@@ -0,0 +1,27 @@
+//
+//  loadScriptFromMessage.js
+//  examples
+//
+//  Created by Thijs Wenker on 9/15/14.
+//  Copyright 2014 High Fidelity, Inc.
+//  
+//  Filters script links out of incomming messages and prompts you to run the script.
+//
+//  Distributed under the Apache License, Version 2.0.
+//  See the accompanying file LICENSE or http://www.apache.org/licenses/LICENSE-2.0.html
+//
+
+//Javascript link RegEX
+const JS_LINK_REGEX = /https?:\/\/[^ ]+\.js/i;
+
+function onIncomingMessage(user, message) {
+    var script_link = JS_LINK_REGEX.exec(message);
+    if (script_link == null) {
+        return;
+    }
+    if (Window.confirm("@" + user + " sent the following script:\n" + script_link + "\nwould you like to run it?")) {
+        Script.load(script_link);
+    }
+}
+
+GlobalServices.incomingMessage.connect(onIncomingMessage);

interface/src/Audio.cpp

@@ -506,30 +506,34 @@ void Audio::handleAudioInput() {
 
     QByteArray inputByteArray = _inputDevice->readAll();
 
-    int16_t* inputFrameData = (int16_t*)inputByteArray.data();
-    const int inputFrameCount = inputByteArray.size() / sizeof(int16_t);
+    if (!_muted && (_audioSourceInjectEnabled || _peqEnabled)) {
+        
+        int16_t* inputFrameData = (int16_t*)inputByteArray.data();
+        const int inputFrameCount = inputByteArray.size() / sizeof(int16_t);
 
-    _inputFrameBuffer.copyFrames(1, inputFrameCount, inputFrameData, false /*copy in*/);
+        _inputFrameBuffer.copyFrames(1, inputFrameCount, inputFrameData, false /*copy in*/);
 
- //   _inputGain.render(_inputFrameBuffer);  // input/mic gain+mute
-    
-    //  Add audio source injection if enabled
-    if (_audioSourceInjectEnabled && !_muted) {
-     
-        if (_toneSourceEnabled) {  // sine generator
-            _toneSource.render(_inputFrameBuffer);
+#if ENABLE_INPUT_GAIN
+        _inputGain.render(_inputFrameBuffer);  // input/mic gain+mute
+#endif
+        //  Add audio source injection if enabled
+        if (_audioSourceInjectEnabled) {
+         
+            if (_toneSourceEnabled) {  // sine generator
+                _toneSource.render(_inputFrameBuffer);
+            }
+            else if(_noiseSourceEnabled) { // pink noise generator
+                _noiseSource.render(_inputFrameBuffer);
+            }
+            _sourceGain.render(_inputFrameBuffer); // post gain
         }
-        else if(_noiseSourceEnabled) { // pink noise generator
-            _noiseSource.render(_inputFrameBuffer);
+        if (_peqEnabled) {
+            _peq.render(_inputFrameBuffer); // 3-band parametric eq
         }
-        _sourceGain.render(_inputFrameBuffer); // post gain
-    }
-    if (_peqEnabled && !_muted) {
-        _peq.render(_inputFrameBuffer); // 3-band parametric eq
-    }
 
-    _inputFrameBuffer.copyFrames(1, inputFrameCount, inputFrameData, true /*copy out*/);
-    
+        _inputFrameBuffer.copyFrames(1, inputFrameCount, inputFrameData, true /*copy out*/);
+    }
+        
     if (Menu::getInstance()->isOptionChecked(MenuOption::EchoLocalAudio) && !_muted && _audioOutput) {
         // if this person wants local loopback add that to the locally injected audio
 

libraries/audio/src/AudioSourceTone.cpp

@@ -17,4 +17,43 @@
 #include "AudioBuffer.h"
 #include "AudioSourceTone.h"
 
-uint32_t AudioSourceTone::_frameOffset = 0;
+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

@@ -12,61 +12,52 @@
 #ifndef hifi_AudioSourceTone_h
 #define hifi_AudioSourceTone_h
 
+// Implements a two-pole Gordon-Smith oscillator
 class AudioSourceTone
 {
-    static uint32_t _frameOffset;
     float32_t _frequency;
     float32_t _amplitude;
     float32_t _sampleRate;
     float32_t _omega;
+    float32_t _epsilon;
+    float32_t _yq1;  
+    float32_t _y1;
+    
+    void updateCoefficients();
     
 public:
-    AudioSourceTone() {
-        initialize();
-    }
+    AudioSourceTone();
+    ~AudioSourceTone();
     
-    ~AudioSourceTone() {
-        finalize();
-    }
-    
-    void initialize() {
-        _frameOffset = 0;
-        setParameters(SAMPLE_RATE, 220.0f, 0.9f);
-    }
-    
-    void finalize() {
-    }
-    
-    void reset() {
-        _frameOffset = 0;
-    }
+    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);
-        _omega = _frequency / _sampleRate * TWO_PI;
-    }
+    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) {
-        
-        // note: this is a placeholder implementation. final version will not include any transcendental ops in our render loop
-        
-        float32_t** samples = frameBuffer.getFrameData();
-        for (uint16_t i = 0; i < frameBuffer.getFrameCount(); ++i) {
-            for (uint16_t j = 0; j < frameBuffer.getChannelCount(); ++j) {
-                samples[j][i] = sinf((i + _frameOffset) * _omega);
-            }
-        }
-        _frameOffset += frameBuffer.getFrameCount();
-    }
+    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) {
+        
+        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