diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index 97658c720f..7e9128551a 100644
--- a/webrtc/common_types.h
+++ b/webrtc/common_types.h
@@ -342,13 +342,6 @@ typedef struct        // All levels are reported in dB
     StatVal a_nlp;
 } EchoStatistics;
 
-enum TelephoneEventDetectionMethods
-{
-    kInBand = 0,
-    kOutOfBand = 1,
-    kInAndOutOfBand = 2
-};
-
 enum NsModes    // type of Noise Suppression
 {
     kNsUnchanged = 0,   // previously set mode
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
index 7fcef2099c..b436e3ff99 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -13,8 +13,8 @@
 
 #include <vector>
 
-#include "modules/interface/module.h"
-#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 
 namespace webrtc {
 // Forward declarations.
@@ -796,19 +796,11 @@ class RtpRtcp : public Module {
         const WebRtc_UWord16 packetSizeSamples) = 0;
 
     /*
-    *   Outband TelephoneEvent(DTMF) detection
+    *   Forward DTMF to decoder for playout.
     *
     *   return -1 on failure else 0
     */
-    virtual WebRtc_Word32 SetTelephoneEventStatus(
-        const bool enable,
-        const bool forwardToDecoder,
-        const bool detectEndOfTone = false) = 0;
-
-    /*
-    *   Is outband TelephoneEvent(DTMF) turned on/off?
-    */
-    virtual bool TelephoneEvent() const = 0;
+    virtual int SetTelephoneEventForwardToDecoder(bool forwardToDecoder) = 0;
 
     /*
     *   Returns true if received DTMF events are forwarded to the decoder using
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
index 38891999dd..2b13d39518 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
@@ -207,9 +207,6 @@ protected:
 
 class RtpAudioFeedback {
  public:
-  virtual void OnReceivedTelephoneEvent(const WebRtc_Word32 id,
-                                        const WebRtc_UWord8 event,
-                                        const bool endOfEvent) = 0;
 
   virtual void OnPlayTelephoneEvent(const WebRtc_Word32 id,
                                     const WebRtc_UWord8 event,
@@ -304,10 +301,6 @@ class NullRtpAudioFeedback : public RtpAudioFeedback {
  public:
   virtual ~NullRtpAudioFeedback() {}
 
-  virtual void OnReceivedTelephoneEvent(const WebRtc_Word32 id,
-                                        const WebRtc_UWord8 event,
-                                        const bool endOfEvent) {}
-
   virtual void OnPlayTelephoneEvent(const WebRtc_Word32 id,
                                     const WebRtc_UWord8 event,
                                     const WebRtc_UWord16 lengthMs,
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index e4f3c1c83a..74fbda7889 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -247,10 +247,7 @@ class MockRtpRtcp : public RtpRtcp {
       WebRtc_Word32(RtpAudioFeedback* messagesCallback));
   MOCK_METHOD1(SetAudioPacketSize,
       WebRtc_Word32(const WebRtc_UWord16 packetSizeSamples));
-  MOCK_METHOD3(SetTelephoneEventStatus,
-      WebRtc_Word32(const bool enable, const bool forwardToDecoder, const bool detectEndOfTone));
-  MOCK_CONST_METHOD0(TelephoneEvent,
-      bool());
+  MOCK_METHOD1(SetTelephoneEventForwardToDecoder, int(bool forwardToDecoder));
   MOCK_CONST_METHOD0(TelephoneEventForwardToDecoder,
       bool());
   MOCK_CONST_METHOD1(SendTelephoneEventActive,
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index 50847f2841..1ae0a7dd3f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -27,9 +27,7 @@ RTPReceiverAudio::RTPReceiverAudio(const WebRtc_Word32 id,
       critical_section_rtp_receiver_audio_(
           CriticalSectionWrapper::CreateCriticalSection()),
       last_received_frequency_(8000),
-      telephone_event_(false),
       telephone_event_forward_to_decoder_(false),
-      telephone_event_detect_end_of_tone_(false),
       telephone_event_payload_type_(-1),
       cng_nb_payload_type_(-1),
       cng_wb_payload_type_(-1),
@@ -51,23 +49,13 @@ WebRtc_UWord32 RTPReceiverAudio::AudioFrequency() const {
 }
 
 // Outband TelephoneEvent(DTMF) detection
-WebRtc_Word32 RTPReceiverAudio::SetTelephoneEventStatus(
-    const bool enable,
-    const bool forward_to_decoder,
-    const bool detect_end_of_tone) {
+int RTPReceiverAudio::SetTelephoneEventForwardToDecoder(
+    bool forward_to_decoder) {
   CriticalSectionScoped lock(critical_section_rtp_receiver_audio_.get());
-  telephone_event_ = enable;
-  telephone_event_detect_end_of_tone_ = detect_end_of_tone;
   telephone_event_forward_to_decoder_ = forward_to_decoder;
   return 0;
 }
 
-// Is outband TelephoneEvent(DTMF) turned on/off?
-bool RTPReceiverAudio::TelephoneEvent() const {
-  CriticalSectionScoped lock(critical_section_rtp_receiver_audio_.get());
-  return telephone_event_;
-}
-
 // Is forwarding of outband telephone events turned on/off?
 bool RTPReceiverAudio::TelephoneEventForwardToDecoder() const {
   CriticalSectionScoped lock(critical_section_rtp_receiver_audio_.get());
@@ -194,34 +182,6 @@ WebRtc_Word32 RTPReceiverAudio::OnNewPayloadTypeCreated(
   return 0;
 }
 
-void RTPReceiverAudio::SendTelephoneEvents(
-    WebRtc_UWord8 number_of_new_events,
-    WebRtc_UWord8 new_events[MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS],
-    WebRtc_UWord8 number_of_removed_events,
-    WebRtc_UWord8 removed_events[MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS]) {
-
-  // Copy these variables since we can't hold the critsect when we call the
-  // callback. cb_audio_feedback_ and id_ are immutable though.
-  bool telephone_event;
-  bool telephone_event_detect_end_of_tone;
-  {
-    CriticalSectionScoped lock(critical_section_rtp_receiver_audio_.get());
-    telephone_event = telephone_event_;
-    telephone_event_detect_end_of_tone = telephone_event_detect_end_of_tone_;
-  }
-  if (telephone_event) {
-    for (int n = 0; n < number_of_new_events; ++n) {
-      cb_audio_feedback_->OnReceivedTelephoneEvent(id_, new_events[n], false);
-    }
-    if (telephone_event_detect_end_of_tone) {
-      for (int n = 0; n < number_of_removed_events; ++n) {
-        cb_audio_feedback_->OnReceivedTelephoneEvent(
-            id_, removed_events[n], true);
-      }
-    }
-  }
-}
-
 WebRtc_Word32 RTPReceiverAudio::ParseRtpPacket(
     WebRtcRTPHeader* rtp_header,
     const ModuleRTPUtility::PayloadUnion& specific_payload,
@@ -316,10 +276,6 @@ WebRtc_Word32 RTPReceiverAudio::ParseAudioCodecSpecific(
     const WebRtc_UWord16 payload_length,
     const ModuleRTPUtility::AudioPayload& audio_specific,
     const bool is_red) {
-  WebRtc_UWord8 new_events[MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS] = {};
-  WebRtc_UWord8 removed_events[MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS] = {};
-  WebRtc_UWord8 number_of_new_events = 0;
-  WebRtc_UWord8 number_of_removed_events = 0;
 
   if (payload_length == 0) {
     return 0;
@@ -355,16 +311,12 @@ WebRtc_Word32 RTPReceiverAudio::ParseAudioCodecSpecific(
       if (event != telephone_event_reported_.end()) {
         // we have already seen this event
         if (end) {
-          removed_events[number_of_removed_events] = payload_data[4 * n];
-          number_of_removed_events++;
           telephone_event_reported_.erase(payload_data[4 * n]);
         }
       } else {
         if (end) {
           // don't add if it's a end of a tone
         } else {
-          new_events[number_of_new_events] = payload_data[4 * n];
-          number_of_new_events++;
           telephone_event_reported_.insert(payload_data[4 * n]);
         }
       }
@@ -376,12 +328,6 @@ WebRtc_Word32 RTPReceiverAudio::ParseAudioCodecSpecific(
     // RFC 4733 See 2.5.1.5. & 2.5.2.4.  Multiple Events in a Packet
   }
 
-  // This needs to be called without locks held.
-  SendTelephoneEvents(number_of_new_events,
-                      new_events,
-                      number_of_removed_events,
-                      removed_events);
-
   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_audio_.get());
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
index 1a1826dc4e..96df5f1776 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -33,13 +33,8 @@ class RTPReceiverAudio : public RTPReceiverStrategy {
 
   WebRtc_UWord32 AudioFrequency() const;
 
-  // Outband TelephoneEvent (DTMF) detection
-  WebRtc_Word32 SetTelephoneEventStatus(const bool enable,
-                                        const bool forward_to_decoder,
-                                        const bool detect_end_of_tone);
-
-  // Is outband DTMF(AVT) turned on/off?
-  bool TelephoneEvent() const;
+  // Forward DTMFs to decoder for playout.
+  int SetTelephoneEventForwardToDecoder(bool forward_to_decoder);
 
   // Is forwarding of outband telephone events turned on/off?
   bool TelephoneEventForwardToDecoder() const;
@@ -98,11 +93,6 @@ class RTPReceiverAudio : public RTPReceiverStrategy {
                            bool* should_discard_changes);
 
  private:
-  void SendTelephoneEvents(
-      WebRtc_UWord8 number_of_new_events,
-      WebRtc_UWord8 new_events[MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS],
-      WebRtc_UWord8 number_of_removed_events,
-      WebRtc_UWord8 removed_events[MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS]);
 
   WebRtc_Word32 ParseAudioCodecSpecific(
       WebRtcRTPHeader* rtp_header,
@@ -116,9 +106,7 @@ class RTPReceiverAudio : public RTPReceiverStrategy {
 
   WebRtc_UWord32 last_received_frequency_;
 
-  bool telephone_event_;
   bool telephone_event_forward_to_decoder_;
-  bool telephone_event_detect_end_of_tone_;
   WebRtc_Word8 telephone_event_payload_type_;
   std::set<WebRtc_UWord8> telephone_event_reported_;
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 02d1e14d3e..c463f80bfa 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -1582,29 +1582,17 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetStorePacketsStatus(
   return 0;  // TODO(pwestin): change to void.
 }
 
-// Out-band TelephoneEvent detection.
-WebRtc_Word32 ModuleRtpRtcpImpl::SetTelephoneEventStatus(
-    const bool enable,
-    const bool forward_to_decoder,
-    const bool detect_end_of_tone) {
+// Forward DTMFs to decoder for playout.
+int ModuleRtpRtcpImpl::SetTelephoneEventForwardToDecoder(
+    bool forward_to_decoder) {
   WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-               "SetTelephoneEventStatus(enable:%d forward_to_decoder:%d"
-               " detect_end_of_tone:%d)", enable, forward_to_decoder,
-               detect_end_of_tone);
+               "SetTelephoneEventForwardToDecoder(forward_to_decoder:%d)",
+               forward_to_decoder);
 
   assert(audio_);
   assert(rtp_telephone_event_handler_);
-  return rtp_telephone_event_handler_->SetTelephoneEventStatus(
-           enable, forward_to_decoder, detect_end_of_tone);
-}
-
-// Is out-band TelephoneEvent turned on/off?
-bool ModuleRtpRtcpImpl::TelephoneEvent() const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "TelephoneEvent()");
-
-  assert(audio_);
-  assert(rtp_telephone_event_handler_);
-  return rtp_telephone_event_handler_->TelephoneEvent();
+  return rtp_telephone_event_handler_->SetTelephoneEventForwardToDecoder(
+      forward_to_decoder);
 }
 
 // Is forwarding of out-band telephone events turned on/off?
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index a49ca3a570..c8283238ac 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -345,14 +345,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   virtual WebRtc_Word32 SetAudioPacketSize(
       const WebRtc_UWord16 packet_size_samples);
 
-  // Outband DTMF detection.
-  virtual WebRtc_Word32 SetTelephoneEventStatus(
-      const bool enable,
-      const bool forward_to_decoder,
-      const bool detect_end_of_tone = false);
-
-  // Is outband DTMF turned on/off?
-  virtual bool TelephoneEvent() const;
+  // Forward DTMFs to decoder for playout.
+  virtual int SetTelephoneEventForwardToDecoder(bool forward_to_decoder);
 
   // Is forwarding of outband telephone events turned on/off?
   virtual bool TelephoneEventForwardToDecoder() const;
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 2c0c2c2a4e..f59b192fe3 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -190,11 +190,8 @@ TEST_F(RtpRtcpAudioTest, Basic) {
   EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
   EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
 
-  EXPECT_FALSE(module1->TelephoneEvent());
-
   // Test detection at the end of a DTMF tone.
-  EXPECT_EQ(0, module2->SetTelephoneEventStatus(true, true, true));
-  EXPECT_EQ(true, module2->TelephoneEvent());
+  EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
 
   EXPECT_EQ(0, module1->SetSendingStatus(true));
 
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 56de748b97..1eabee8401 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -93,28 +93,6 @@ Channel::InFrameType(WebRtc_Word16 frameType)
     return 0;
 }
 
-#ifdef WEBRTC_DTMF_DETECTION
-int
-Channel::IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool end)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-               "Channel::IncomingDtmf(digitDtmf=%u, end=%d)",
-               digitDtmf, end);
-
-    if (digitDtmf != 999)
-    {
-        CriticalSectionScoped cs(&_callbackCritSect);
-        if (_telephoneEventDetectionPtr)
-        {
-            _telephoneEventDetectionPtr->OnReceivedTelephoneEventInband(
-                _channelId, digitDtmf, end);
-        }
-    }
-
-    return 0;
-}
-#endif
-
 WebRtc_Word32
 Channel::OnRxVadDetected(const int vadDecision)
 {
@@ -510,29 +488,6 @@ Channel::IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
     }
 }
 
-void
-Channel::OnReceivedTelephoneEvent(const WebRtc_Word32 id,
-                                  const WebRtc_UWord8 event,
-                                  const bool endOfEvent)
-{
-    WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::OnReceivedTelephoneEvent(id=%d, event=%u,"
-                 " endOfEvent=%d)", id, event, endOfEvent);
-
-#ifdef WEBRTC_DTMF_DETECTION
-    if (_outOfBandTelephoneEventDetecion)
-    {
-        CriticalSectionScoped cs(&_callbackCritSect);
-
-        if (_telephoneEventDetectionPtr)
-        {
-            _telephoneEventDetectionPtr->OnReceivedTelephoneEventOutOfBand(
-                _channelId, event, endOfEvent);
-        }
-    }
-#endif
-}
-
 void
 Channel::OnPlayTelephoneEvent(const WebRtc_Word32 id,
                               const WebRtc_UWord8 event,
@@ -1123,9 +1078,6 @@ Channel::Channel(const WebRtc_Word32 channelId,
     _encryptionPtr(NULL),
     _rtpAudioProc(NULL),
     _rxAudioProcessingModulePtr(NULL),
-#ifdef WEBRTC_DTMF_DETECTION
-    _telephoneEventDetectionPtr(NULL),
-#endif
     _rxVadObserverPtr(NULL),
     _oldVadDecision(-1),
     _sendFrameType(0),
@@ -1149,8 +1101,6 @@ Channel::Channel(const WebRtc_Word32 channelId,
     _decrypting(false),
     _playOutbandDtmfEvent(false),
     _playInbandDtmfEvent(false),
-    _inbandTelephoneEventDetection(false),
-    _outOfBandTelephoneEventDetecion(false),
     _extraPayloadType(0),
     _insertExtraRTPPacket(false),
     _extraMarkerBit(false),
@@ -1268,15 +1218,6 @@ Channel::~Channel()
                      "~Channel() failed to de-register VAD callback"
                      " (Audio coding module)");
     }
-#ifdef WEBRTC_DTMF_DETECTION
-    if (_audioCodingModule.RegisterIncomingMessagesCallback(NULL) == -1)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                     VoEId(_instanceId,_channelId),
-                     "~Channel() failed to de-register incoming messages "
-                     "callback (Audio coding module)");
-    }
-#endif
     // De-register modules in process thread
 #ifndef WEBRTC_EXTERNAL_TRANSPORT
     if (_moduleProcessThreadPtr->DeRegisterModule(&_socketTransportModule)
@@ -1380,7 +1321,7 @@ Channel::Init()
     // be transmitted since the Transport object will then be invalid.
 
     const bool rtpRtcpFail =
-        ((_rtpRtcpModule->SetTelephoneEventStatus(false, true, true) == -1) ||
+        ((_rtpRtcpModule->SetTelephoneEventForwardToDecoder(true) == -1) ||
         // RTCP is enabled by default
         (_rtpRtcpModule->SetRTCPStatus(kRtcpCompound) == -1));
     if (rtpRtcpFail)
@@ -1768,10 +1709,8 @@ Channel::StopReceiving()
         }
     }
 #endif
-    bool dtmfDetection = _rtpRtcpModule->TelephoneEvent();
     // Recover DTMF detection status.
-    WebRtc_Word32 ret = _rtpRtcpModule->SetTelephoneEventStatus(dtmfDetection,
-                                                               true, true);
+    WebRtc_Word32 ret = _rtpRtcpModule->SetTelephoneEventForwardToDecoder(true);
     if (ret != 0) {
         _engineStatisticsPtr->SetLastError(
             VE_INVALID_OPERATION, kTraceWarning,
@@ -4455,144 +4394,6 @@ Channel::GetSendTelephoneEventPayloadType(unsigned char& type)
     return 0;
 }
 
-#ifdef WEBRTC_DTMF_DETECTION
-
-WebRtc_Word32
-Channel::RegisterTelephoneEventDetection(
-    TelephoneEventDetectionMethods detectionMethod,
-    VoETelephoneEventObserver& observer)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::RegisterTelephoneEventDetection()");
-    CriticalSectionScoped cs(&_callbackCritSect);
-
-    if (_telephoneEventDetectionPtr)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_INVALID_OPERATION, kTraceError,
-            "RegisterTelephoneEventDetection() detection already enabled");
-        return -1;
-    }
-
-    _telephoneEventDetectionPtr = &observer;
-
-    switch (detectionMethod)
-    {
-        case kInBand:
-            _inbandTelephoneEventDetection = true;
-            _outOfBandTelephoneEventDetecion = false;
-            break;
-        case kOutOfBand:
-            _inbandTelephoneEventDetection = false;
-            _outOfBandTelephoneEventDetecion = true;
-            break;
-        case kInAndOutOfBand:
-            _inbandTelephoneEventDetection = true;
-            _outOfBandTelephoneEventDetecion = true;
-            break;
-        default:
-            _engineStatisticsPtr->SetLastError(
-                VE_INVALID_ARGUMENT, kTraceError,
-                "RegisterTelephoneEventDetection() invalid detection method");
-            return -1;
-    }
-
-    if (_inbandTelephoneEventDetection)
-    {
-        // Enable in-band Dtmf detectin in the ACM.
-        if (_audioCodingModule.RegisterIncomingMessagesCallback(this) != 0)
-        {
-            _engineStatisticsPtr->SetLastError(
-                VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
-                "RegisterTelephoneEventDetection() failed to enable Dtmf "
-                "detection");
-        }
-    }
-
-    // Enable/disable out-of-band detection of received telephone-events.
-    // When enabled, RtpAudioFeedback::OnReceivedTelephoneEvent() will be
-    // called two times by the RTP/RTCP module (start & end).
-    const bool forwardToDecoder =
-        _rtpRtcpModule->TelephoneEventForwardToDecoder();
-    const bool detectEndOfTone = true;
-    _rtpRtcpModule->SetTelephoneEventStatus(_outOfBandTelephoneEventDetecion,
-                                           forwardToDecoder,
-                                           detectEndOfTone);
-
-    return 0;
-}
-
-int
-Channel::DeRegisterTelephoneEventDetection()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "Channel::DeRegisterTelephoneEventDetection()");
-
-    CriticalSectionScoped cs(&_callbackCritSect);
-
-    if (!_telephoneEventDetectionPtr)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_INVALID_OPERATION,
-            kTraceWarning,
-            "DeRegisterTelephoneEventDetection() detection already disabled");
-        return 0;
-    }
-
-    // Disable out-of-band event detection
-    const bool forwardToDecoder =
-        _rtpRtcpModule->TelephoneEventForwardToDecoder();
-    _rtpRtcpModule->SetTelephoneEventStatus(false, forwardToDecoder);
-
-    // Disable in-band Dtmf detection
-    _audioCodingModule.RegisterIncomingMessagesCallback(NULL);
-
-    _inbandTelephoneEventDetection = false;
-    _outOfBandTelephoneEventDetecion = false;
-    _telephoneEventDetectionPtr = NULL;
-
-    return 0;
-}
-
-int
-Channel::GetTelephoneEventDetectionStatus(
-    bool& enabled,
-    TelephoneEventDetectionMethods& detectionMethod)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "Channel::GetTelephoneEventDetectionStatus()");
-
-    {
-        CriticalSectionScoped cs(&_callbackCritSect);
-        enabled = (_telephoneEventDetectionPtr != NULL);
-    }
-
-    if (enabled)
-    {
-        if (_inbandTelephoneEventDetection && !_outOfBandTelephoneEventDetecion)
-            detectionMethod = kInBand;
-        else if (!_inbandTelephoneEventDetection
-            && _outOfBandTelephoneEventDetecion)
-            detectionMethod = kOutOfBand;
-        else if (_inbandTelephoneEventDetection
-            && _outOfBandTelephoneEventDetecion)
-            detectionMethod = kInAndOutOfBand;
-        else
-        {
-            assert(false);
-            return -1;
-        }
-    }
-
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-               VoEId(_instanceId, _channelId),
-               "GetTelephoneEventDetectionStatus() => enabled=%d,"
-               "detectionMethod=%d", enabled, detectionMethod);
-    return 0;
-}
-
-#endif  // #ifdef WEBRTC_DTMF_DETECTION
-
 int
 Channel::UpdateRxVadDetection(AudioFrame& audioFrame)
 {
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 6fdcdabcc4..f1c5809be3 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -73,9 +73,6 @@ class Channel:
     public RtpAudioFeedback,
     public AudioPacketizationCallback, // receive encoded packets from the ACM
     public ACMVADCallback, // receive voice activity from the ACM
-#ifdef WEBRTC_DTMF_DETECTION
-    public AudioCodingFeedback, // inband Dtmf detection in the ACM
-#endif
     public MixerParticipant // supplies output mixer with audio frames
 {
 public:
@@ -296,15 +293,6 @@ public:
     bool DtmfPlayoutStatus() const;
     int SetSendTelephoneEventPayloadType(unsigned char type);
     int GetSendTelephoneEventPayloadType(unsigned char& type);
-#ifdef WEBRTC_DTMF_DETECTION
-    int RegisterTelephoneEventDetection(
-            TelephoneEventDetectionMethods detectionMethod,
-            VoETelephoneEventObserver& observer);
-    int DeRegisterTelephoneEventDetection();
-    int GetTelephoneEventDetectionStatus(
-            bool& enabled,
-            TelephoneEventDetectionMethods& detectionMethod);
-#endif
 
     // VoEAudioProcessingImpl
     int UpdateRxVadDetection(AudioFrame& audioFrame);
@@ -372,11 +360,6 @@ public:
     // From ACMVADCallback in the ACM
     WebRtc_Word32 InFrameType(WebRtc_Word16 frameType);
 
-#ifdef WEBRTC_DTMF_DETECTION
-public: // From AudioCodingFeedback in the ACM
-    int IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool end);
-#endif
-
 public:
     WebRtc_Word32 OnRxVadDetected(const int vadDecision);
 
@@ -603,9 +586,6 @@ private:
     Encryption* _encryptionPtr; // WebRtc SRTP or external encryption
     scoped_ptr<AudioProcessing> _rtpAudioProc;
     AudioProcessing* _rxAudioProcessingModulePtr; // far end AudioProcessing
-#ifdef WEBRTC_DTMF_DETECTION
-    VoETelephoneEventObserver* _telephoneEventDetectionPtr;
-#endif
     VoERxVadCallback* _rxVadObserverPtr;
     WebRtc_Word32 _oldVadDecision;
     WebRtc_Word32 _sendFrameType; // Send data is voice, 1-voice, 0-otherwise
@@ -634,8 +614,6 @@ private:
     // VoEDtmf
     bool _playOutbandDtmfEvent;
     bool _playInbandDtmfEvent;
-    bool _inbandTelephoneEventDetection;
-    bool _outOfBandTelephoneEventDetecion;
     // VoeRTP_RTCP
     WebRtc_UWord8 _extraPayloadType;
     bool _insertExtraRTPPacket;
diff --git a/webrtc/voice_engine/include/voe_dtmf.h b/webrtc/voice_engine/include/voe_dtmf.h
index 3ed174930a..301d73f8d5 100644
--- a/webrtc/voice_engine/include/voe_dtmf.h
+++ b/webrtc/voice_engine/include/voe_dtmf.h
@@ -33,35 +33,12 @@
 #ifndef WEBRTC_VOICE_ENGINE_VOE_DTMF_H
 #define WEBRTC_VOICE_ENGINE_VOE_DTMF_H
 
-#include "common_types.h"
+#include "webrtc/common_types.h"
 
 namespace webrtc {
 
 class VoiceEngine;
 
-// VoETelephoneEventObserver
-class WEBRTC_DLLEXPORT VoETelephoneEventObserver
-{
-public:
-    // This method will be called after the detection of an inband
-    // telephone event. The event code is given as output in the
-    // |eventCode| parameter.
-    virtual void OnReceivedTelephoneEventInband(int channel,
-                                                int eventCode,
-                                                bool endOfEvent) = 0;
-
-    // This method will be called after the detection of an out-of-band
-    // telephone event. The event code is given as output in the
-    // |eventCode| parameter.
-    virtual void OnReceivedTelephoneEventOutOfBand(
-        int channel,
-        int eventCode,
-        bool endOfEvent) = 0;
-
-protected:
-    virtual ~VoETelephoneEventObserver() {}
-};
-
 // VoEDtmf
 class WEBRTC_DLLEXPORT VoEDtmf
 {
@@ -122,22 +99,6 @@ public:
     // Stops playing out a DTMF feedback tone locally.
     virtual int StopPlayingDtmfTone() = 0;
 
-    // Installs an instance of a VoETelephoneEventObserver derived class and
-    // activates detection of telephone events for the specified |channel|.
-    virtual int RegisterTelephoneEventDetection(
-        int channel, TelephoneEventDetectionMethods detectionMethod,
-        VoETelephoneEventObserver& observer) = 0;
-
-    // Removes an instance of a VoETelephoneEventObserver derived class and
-    // disables detection of telephone events for the specified |channel|.
-    virtual int DeRegisterTelephoneEventDetection(int channel) = 0;
-
-    // Gets the current telephone-event detection status for a specified
-    // |channel|.
-    virtual int GetTelephoneEventDetectionStatus(
-        int channel, bool& enabled,
-        TelephoneEventDetectionMethods& detectionMethod) = 0;
-
 protected:
     VoEDtmf() {}
     virtual ~VoEDtmf() {}
diff --git a/webrtc/voice_engine/test/auto_test/voe_extended_test.cc b/webrtc/voice_engine/test/auto_test/voe_extended_test.cc
index 2c9088e653..d4c49c449e 100644
--- a/webrtc/voice_engine/test/auto_test/voe_extended_test.cc
+++ b/webrtc/voice_engine/test/auto_test/voe_extended_test.cc
@@ -3319,71 +3319,6 @@ int VoEExtendedTest::TestDtmf() {
   AOK();
   ANL();
 
-#ifdef WEBRTC_DTMF_DETECTION
-  TEST(RegisterTelephoneEventDetection - several channels); ANL();
-
-  ci.channels = 1;
-  ci.pacsize = 160;
-  ci.plfreq = 8000;
-  ci.pltype = 0;
-  ci.rate = 64000;
-  strcpy(ci.plname, "PCMU");
-  TEST_MUSTPASS(codec->SetSendCodec(0, ci));
-
-  int ch2 = voe_base_->CreateChannel();
-  TEST_MUSTPASS(voe_base_->SetSendDestination(ch2, 8002, "127.0.0.1"));
-  TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch2, 8002));
-  TEST_MUSTPASS(voe_base_->StartReceive(ch2));
-  TEST_MUSTPASS(codec->SetSendCodec(ch2, ci));
-  TEST_MUSTPASS(voe_base_->StartPlayout(ch2));
-  TEST_MUSTPASS(voe_base_->StartSend(ch2));
-  MARK();
-
-  DtmfCallback *d = new DtmfCallback();
-  TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(false));
-
-  TEST_MUSTPASS(voe_base_->StopSend(0));
-  TEST_MUSTPASS(voe_base_->StopPlayout(0));
-  TEST_MUSTPASS(voe_base_->StartSend(0));
-  TEST_MUSTPASS(voe_base_->StartPlayout(0));
-
-  // In-band
-  TEST_MUSTPASS(dtmf->RegisterTelephoneEventDetection(0, kInBand, *d));
-  TEST_MUSTPASS(dtmf->RegisterTelephoneEventDetection(ch2, kInBand, *d));
-  TEST_LOG("\nSending in-band telephone events:");
-  for(int i = 0; i < 16; i++)
-  {
-    TEST_LOG("\n  %d ", i); fflush(NULL);
-    TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, i, false, 160, 10));
-    TEST_MUSTPASS(dtmf->SendTelephoneEvent(ch2, i, false, 160, 10));
-    SleepMs(500);
-  }
-  TEST_LOG("\nDetected %d events \n", d->counter);
-  TEST_MUSTPASS(d->counter != 32);
-  TEST_MUSTPASS(dtmf->DeRegisterTelephoneEventDetection(0));
-  TEST_MUSTPASS(dtmf->DeRegisterTelephoneEventDetection(ch2));
-
-  // Out-of-band
-  d->counter = 0;
-  TEST_MUSTPASS(dtmf->RegisterTelephoneEventDetection(0, kOutOfBand, *d));
-  TEST_MUSTPASS(dtmf->RegisterTelephoneEventDetection(ch2, kOutOfBand, *d));
-  TEST_LOG("\nSending out-band telephone events:");
-  for(int i = 0; i < 16; i++)
-  {
-    TEST_LOG("\n  %d ", i); fflush(NULL);
-    TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, i, true, 160, 10));
-    TEST_MUSTPASS(dtmf->SendTelephoneEvent(ch2, i, true, 160, 10));
-    SleepMs(500);
-  }
-  TEST_LOG("\nDetected %d events \n", d->counter);
-  TEST_MUSTPASS(d->counter != 32);
-  TEST_MUSTPASS(dtmf->DeRegisterTelephoneEventDetection(0));
-  TEST_MUSTPASS(dtmf->DeRegisterTelephoneEventDetection(ch2));
-  delete d;
-
-  AOK(); ANL();
-#endif
-
   TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(true, false));
   TEST_MUSTPASS(voe_base_->StopSend(0));
   TEST_MUSTPASS(voe_base_->StopPlayout(0));
diff --git a/webrtc/voice_engine/voe_dtmf_impl.cc b/webrtc/voice_engine/voe_dtmf_impl.cc
index 5353cf413e..4d1a7a939a 100644
--- a/webrtc/voice_engine/voe_dtmf_impl.cc
+++ b/webrtc/voice_engine/voe_dtmf_impl.cc
@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "voe_dtmf_impl.h"
+#include "webrtc/voice_engine/voe_dtmf_impl.h"
 
-#include "channel.h"
-#include "critical_section_wrapper.h"
-#include "output_mixer.h"
-#include "trace.h"
-#include "transmit_mixer.h"
-#include "voe_errors.h"
-#include "voice_engine_impl.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/voice_engine/channel.h"
+#include "webrtc/voice_engine/include/voe_errors.h"
+#include "webrtc/voice_engine/output_mixer.h"
+#include "webrtc/voice_engine/transmit_mixer.h"
+#include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 
@@ -271,94 +271,6 @@ int VoEDtmfImpl::StopPlayingDtmfTone()
     return _shared->output_mixer()->StopPlayingDtmfTone();
 }
 
-int VoEDtmfImpl::RegisterTelephoneEventDetection(
-    int channel,
-    TelephoneEventDetectionMethods detectionMethod,
-    VoETelephoneEventObserver& observer)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-      "RegisterTelephoneEventDetection(channel=%d, detectionMethod=%d,"
-      "observer=0x%x)", channel, detectionMethod, &observer);
-#ifdef WEBRTC_DTMF_DETECTION
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ScopedChannel sc(_shared->channel_manager(), channel);
-    voe::Channel* channelPtr = sc.ChannelPtr();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "RegisterTelephoneEventDetection() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->RegisterTelephoneEventDetection(detectionMethod,
-                                                       observer);
-#else
-    _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-        "SetTelephoneEventDetectionStatus() Dtmf detection is not supported");
-    return -1;
-#endif
-}
-
-int VoEDtmfImpl::DeRegisterTelephoneEventDetection(int channel)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-            "DeRegisterTelephoneEventDetection(channel=%d)", channel);
-#ifdef WEBRTC_DTMF_DETECTION
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ScopedChannel sc(_shared->channel_manager(), channel);
-    voe::Channel* channelPtr = sc.ChannelPtr();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "DeRegisterTelephoneEventDe tection() failed to locate channel");
-            return -1;
-    }
-    return channelPtr->DeRegisterTelephoneEventDetection();
-#else
-    _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-        "DeRegisterTelephoneEventDetection() Dtmf detection is not supported");
-    return -1;
-#endif
-}
-
-
-int VoEDtmfImpl::GetTelephoneEventDetectionStatus(
-    int channel,
-    bool& enabled,
-    TelephoneEventDetectionMethods& detectionMethod)
-{
-    WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetTelephoneEventDetectionStatus(channel=%d)", channel);
-#ifdef WEBRTC_DTMF_DETECTION
-    if (!_shared->statistics().Initialized())
-    {
-        _shared->SetLastError(VE_NOT_INITED, kTraceError);
-        return -1;
-    }
-    voe::ScopedChannel sc(_shared->channel_manager(), channel);
-    voe::Channel* channelPtr = sc.ChannelPtr();
-    if (channelPtr == NULL)
-    {
-        _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-            "GetTelephoneEventDetectionStatus() failed to locate channel");
-        return -1;
-    }
-    return channelPtr->GetTelephoneEventDetectionStatus(enabled,
-                                                        detectionMethod);
-#else
-    _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
-        "GetTelephoneEventDetectionStatus() Dtmf detection is not supported");
-    return -1;
-#endif
-}
-
 int VoEDtmfImpl::SetDtmfFeedbackStatus(bool enable, bool directFeedback)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
diff --git a/webrtc/voice_engine/voe_dtmf_impl.h b/webrtc/voice_engine/voe_dtmf_impl.h
index ad3874bf18..e779b69b26 100644
--- a/webrtc/voice_engine/voe_dtmf_impl.h
+++ b/webrtc/voice_engine/voe_dtmf_impl.h
@@ -11,9 +11,8 @@
 #ifndef WEBRTC_VOICE_ENGINE_VOE_DTMF_IMPL_H
 #define WEBRTC_VOICE_ENGINE_VOE_DTMF_IMPL_H
 
-#include "voe_dtmf.h"
-
-#include "shared_data.h"
+#include "webrtc/voice_engine/include/voe_dtmf.h"
+#include "webrtc/voice_engine/shared_data.h"
 
 namespace webrtc
 {
@@ -48,18 +47,6 @@ public:
 
     virtual int StopPlayingDtmfTone();
 
-    virtual int RegisterTelephoneEventDetection(
-        int channel,
-        TelephoneEventDetectionMethods detectionMethod,
-        VoETelephoneEventObserver& observer);
-
-    virtual int DeRegisterTelephoneEventDetection(int channel);
-
-    virtual int GetTelephoneEventDetectionStatus(
-        int channel,
-        bool& enabled,
-        TelephoneEventDetectionMethods& detectionMethod);
-
     virtual int SetDtmfPlayoutStatus(int channel, bool enable);
 
     virtual int GetDtmfPlayoutStatus(int channel, bool& enabled);