Adding one parameter to typing detection tuning

Review URL: https://webrtc-codereview.appspot.com/569009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2203 4adac7df-926f-26a2-2b94-8c16560cd09d

src/voice_engine/main/interface/voe_audio_processing.h

@@ -200,11 +200,13 @@ public:
 
     // Optional setting of typing detection parameters
     // Parameter with value == 0 will be ignored
+
     // and left with default config.
     virtual int SetTypingDetectionParameters(int timeWindow,
                                              int costPerTyping,
                                              int reportingThreshold,
-                                             int penaltyDecay) = 0;
+                                             int penaltyDecay,
+                                             int typeEventDelay) = 0;
 
     // Swaps the capture-side left and right audio channels when enabled. It
     // only has an effect when using a stereo send codec. The setting is

src/voice_engine/main/source/transmit_mixer.cc

@@ -190,6 +190,7 @@ TransmitMixer::TransmitMixer(const WebRtc_UWord32 instanceId) :
     _costPerTyping(100), // Penalty added for a typing + activity coincide
     _reportingThreshold(300), // Threshold for _penaltyCounter
     _penaltyDecay(1), // how much we reduce _penaltyCounter every 10 ms.
+    _typeEventDelay(2), // how "old" event we check for
 #endif
     _saturationWarning(0),
     _noiseWarning(0),
@@ -1373,6 +1374,7 @@ WebRtc_Word32 TransmitMixer::APMProcessStream(
 #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
 int TransmitMixer::TypingDetection()
 {
+
     // We let the VAD determine if we're using this feature or not.
     if (_audioFrame.vad_activity_ == AudioFrame::kVadUnknown)
     {
@@ -1401,7 +1403,8 @@ int TransmitMixer::TypingDetection()
       ++_timeSinceLastTyping;
     }
 
-    if (keyPressed && (_audioFrame.vad_activity_ == AudioFrame::kVadActive)
+    if ((_timeSinceLastTyping < _typeEventDelay)
+        && (_audioFrame.vad_activity_ == AudioFrame::kVadActive)
         && (_timeActive < _timeWindow))
     {
         _penaltyCounter += _costPerTyping;
@@ -1452,7 +1455,8 @@ int TransmitMixer::TimeSinceLastTyping(int &seconds)
 int TransmitMixer::SetTypingDetectionParameters(int timeWindow,
                                                 int costPerTyping,
                                                 int reportingThreshold,
-                                                int penaltyDecay)
+                                                int penaltyDecay,
+                                                int typeEventDelay)
 {
   if(timeWindow != 0)
     _timeWindow = timeWindow;
@@ -1462,6 +1466,9 @@ int TransmitMixer::SetTypingDetectionParameters(int timeWindow,
     _reportingThreshold = reportingThreshold;
   if(penaltyDecay != 0)
     _penaltyDecay = penaltyDecay;
+  if(_typeEventDelay != 0)
+    _penaltyDecay = _typeEventDelay;
+
 
   return(0);
 }

src/voice_engine/main/source/transmit_mixer.h

@@ -154,7 +154,8 @@ public:
     int SetTypingDetectionParameters(int timeWindow,
                                      int costPerTyping,
                                      int reportingThreshold,
-                                     int penaltyDecay);
+                                     int penaltyDecay,
+                                     int typeEventDelay);
 #endif
 
   void EnableStereoChannelSwapping(bool enable);
@@ -217,6 +218,7 @@ private:
     int _costPerTyping; // Penalty added for a typing + activity coincide.
     int _reportingThreshold; // Threshold for _penaltyCounter.
     int _penaltyDecay; // How much we reduce _penaltyCounter every 10 ms.
+    int _typeEventDelay; // How old typing events we allow
 
 #endif
     WebRtc_UWord32 _saturationWarning;

src/voice_engine/main/source/voe_audio_processing_impl.cc

@@ -1077,7 +1077,8 @@ int VoEAudioProcessingImpl::TimeSinceLastTyping(int &seconds) {
 int VoEAudioProcessingImpl::SetTypingDetectionParameters(int timeWindow,
                                                          int costPerTyping,
                                                          int reportingThreshold,
-                                                         int penaltyDecay) {
+                                                         int penaltyDecay,
+                                                         int typeEventDelay) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetTypingDetectionParameters()");
   ANDROID_NOT_SUPPORTED(_shared->statistics());
@@ -1089,7 +1090,7 @@ int VoEAudioProcessingImpl::SetTypingDetectionParameters(int timeWindow,
     return -1;
   }
   return (_shared->transmit_mixer()->SetTypingDetectionParameters(timeWindow,
-      costPerTyping, reportingThreshold, penaltyDecay));
+      costPerTyping, reportingThreshold, penaltyDecay, typeEventDelay));
 
 #else
   _shared->statistics().SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,

src/voice_engine/main/source/voe_audio_processing_impl.h

@@ -91,7 +91,8 @@ class VoEAudioProcessingImpl : public VoEAudioProcessing {
   virtual int SetTypingDetectionParameters(int timeWindow,
                                            int costPerTyping,
                                            int reportingThreshold,
-                                           int penaltyDecay);
+                                           int penaltyDecay,
+                                           int typeEventDelay);
 
   virtual void EnableStereoChannelSwapping(bool enable);
   virtual bool IsStereoChannelSwappingEnabled();