diff --git a/webrtc/media/base/fakemediaengine.h b/webrtc/media/base/fakemediaengine.h
index 2b774b2bb9..d9a79ccca2 100644
--- a/webrtc/media/base/fakemediaengine.h
+++ b/webrtc/media/base/fakemediaengine.h
@@ -304,7 +304,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
   };
   explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine,
                                  const AudioOptions& options)
-      : engine_(engine), time_since_last_typing_(-1), max_bps_(-1) {
+      : engine_(engine), max_bps_(-1) {
     output_scalings_[0] = 1.0;  // For default channel.
     SetOptions(options);
   }
@@ -366,11 +366,6 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
 
   virtual bool GetActiveStreams(AudioInfo::StreamList* streams) { return true; }
   virtual int GetOutputLevel() { return 0; }
-  void set_time_since_last_typing(int ms) { time_since_last_typing_ = ms; }
-  virtual int GetTimeSinceLastTyping() { return time_since_last_typing_; }
-  virtual void SetTypingDetectionParameters(
-      int time_window, int cost_per_typing, int reporting_threshold,
-      int penalty_decay, int type_event_delay) {}
 
   virtual bool CanInsertDtmf() {
     for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
@@ -488,7 +483,6 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
   std::vector<AudioCodec> send_codecs_;
   std::map<uint32_t, double> output_scalings_;
   std::vector<DtmfInfo> dtmf_info_queue_;
-  int time_since_last_typing_;
   AudioOptions options_;
   std::map<uint32_t, VoiceChannelAudioSink*> local_sinks_;
   std::unique_ptr<webrtc::AudioSinkInterface> sink_;
diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
index 6e4011145b..ca02c72fcd 100644
--- a/webrtc/media/base/mediachannel.h
+++ b/webrtc/media/base/mediachannel.h
@@ -991,12 +991,6 @@ class VoiceMediaChannel : public MediaChannel {
   virtual bool GetActiveStreams(AudioInfo::StreamList* actives) = 0;
   // Get the current energy level of the stream sent to the speaker.
   virtual int GetOutputLevel() = 0;
-  // Get the time in milliseconds since last recorded keystroke, or negative.
-  virtual int GetTimeSinceLastTyping() = 0;
-  // Temporarily exposed field for tuning typing detect options.
-  virtual void SetTypingDetectionParameters(int time_window,
-    int cost_per_typing, int reporting_threshold, int penalty_decay,
-    int type_event_delay) = 0;
   // Set speaker output volume of the specified ssrc.
   virtual bool SetOutputVolume(uint32_t ssrc, double volume) = 0;
   // Returns if the telephone-event has been negotiated.
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index b07e9bb5db..6c097eae4d 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -2405,31 +2405,6 @@ int WebRtcVoiceMediaChannel::GetOutputLevel() {
   return highest;
 }
 
-int WebRtcVoiceMediaChannel::GetTimeSinceLastTyping() {
-  int ret;
-  if (engine()->voe()->processing()->TimeSinceLastTyping(ret) == -1) {
-    // In case of error, log the info and continue
-    LOG_RTCERR0(TimeSinceLastTyping);
-    ret = -1;
-  } else {
-    ret *= 1000;  // We return ms, webrtc returns seconds.
-  }
-  return ret;
-}
-
-void WebRtcVoiceMediaChannel::SetTypingDetectionParameters(int time_window,
-    int cost_per_typing, int reporting_threshold, int penalty_decay,
-    int type_event_delay) {
-  if (engine()->voe()->processing()->SetTypingDetectionParameters(
-          time_window, cost_per_typing,
-          reporting_threshold, penalty_decay, type_event_delay) == -1) {
-    // In case of error, log the info and continue
-    LOG_RTCERR5(SetTypingDetectionParameters, time_window,
-                cost_per_typing, reporting_threshold, penalty_decay,
-                type_event_delay);
-  }
-}
-
 bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (ssrc == 0) {
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index 6c88bbf31b..ea48544924 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -183,12 +183,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   bool RemoveRecvStream(uint32_t ssrc) override;
   bool GetActiveStreams(AudioInfo::StreamList* actives) override;
   int GetOutputLevel() override;
-  int GetTimeSinceLastTyping() override;
-  void SetTypingDetectionParameters(int time_window,
-                                    int cost_per_typing,
-                                    int reporting_threshold,
-                                    int penalty_decay,
-                                    int type_event_delay) override;
   bool SetOutputVolume(uint32_t ssrc, double volume) override;
 
   bool CanInsertDtmf() override;