I know this is ugly, but it helps a lot to quickly update webRTC in Chrome and libJingle.
Review URL: https://webrtc-codereview.appspot.com/596004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2290 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -200,13 +200,13 @@ public:
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// Optional setting of typing detection parameters
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// Parameter with value == 0 will be ignored
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// and left with default config.
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// TODO(niklase) Remove default argument as soon as libJingle is updated!
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virtual int SetTypingDetectionParameters(int timeWindow,
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int costPerTyping,
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int reportingThreshold,
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int penaltyDecay,
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int typeEventDelay) = 0;
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int typeEventDelay = 0) = 0;
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// Swaps the capture-side left and right audio channels when enabled. It
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// only has an effect when using a stereo send codec. The setting is
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@ -1467,7 +1467,7 @@ int TransmitMixer::SetTypingDetectionParameters(int timeWindow,
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if(penaltyDecay != 0)
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_penaltyDecay = penaltyDecay;
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if(_typeEventDelay != 0)
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_penaltyDecay = _typeEventDelay;
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_typeEventDelay = typeEventDelay;
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return(0);
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@ -88,11 +88,12 @@ class VoEAudioProcessingImpl : public VoEAudioProcessing {
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virtual int TimeSinceLastTyping(int &seconds);
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// TODO(niklase) Remove default argument as soon as libJingle is updated!
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virtual int SetTypingDetectionParameters(int timeWindow,
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int costPerTyping,
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int reportingThreshold,
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int penaltyDecay,
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int typeEventDelay);
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int typeEventDelay = 0);
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virtual void EnableStereoChannelSwapping(bool enable);
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virtual bool IsStereoChannelSwappingEnabled();
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