diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 160a818323..24afcbcf58 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -125,8 +125,8 @@ bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   return false;
 }
 
-bool AudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
-                                         uint32_t duration_ms) {
+bool AudioSendStream::SendTelephoneEvent(int payload_type, int event,
+                                         int duration_ms) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) &&
          channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index cf0a19ca4b..d463b3da30 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -40,8 +40,8 @@ class AudioSendStream final : public webrtc::AudioSendStream {
   bool DeliverRtcp(const uint8_t* packet, size_t length) override;
 
   // webrtc::AudioSendStream implementation.
-  bool SendTelephoneEvent(int payload_type, uint8_t event,
-                          uint32_t duration_ms) override;
+  bool SendTelephoneEvent(int payload_type, int event,
+                          int duration_ms) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
   const webrtc::AudioSendStream::Config& config() const;
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 6788699cd5..c04a3de77c 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -46,8 +46,8 @@ const CallStatistics kCallStats = {
 const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, 0, -671};
 const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
 const int kTelephoneEventPayloadType = 123;
-const uint8_t kTelephoneEventCode = 45;
-const uint32_t kTelephoneEventDuration = 6789;
+const int kTelephoneEventCode = 45;
+const int kTelephoneEventDuration = 6789;
 
 struct ConfigHelper {
   ConfigHelper()
diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h
index d1af9e0103..24c3d77ab2 100644
--- a/webrtc/audio_send_stream.h
+++ b/webrtc/audio_send_stream.h
@@ -90,8 +90,8 @@ class AudioSendStream : public SendStream {
   };
 
   // TODO(solenberg): Make payload_type a config property instead.
-  virtual bool SendTelephoneEvent(int payload_type, uint8_t event,
-                                  uint32_t duration_ms) = 0;
+  virtual bool SendTelephoneEvent(int payload_type, int event,
+                                  int duration_ms) = 0;
   virtual Stats GetStats() const = 0;
 };
 }  // namespace webrtc
diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
index aa94d48dfb..3277e75a64 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -39,8 +39,8 @@ FakeAudioSendStream::TelephoneEvent
   return latest_telephone_event_;
 }
 
-bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
-                                             uint32_t duration_ms) {
+bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, int event,
+                                             int duration_ms) {
   latest_telephone_event_.payload_type = payload_type;
   latest_telephone_event_.event_code = event;
   latest_telephone_event_.duration_ms = duration_ms;
diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h
index 89a644a296..41a92dfac0 100644
--- a/webrtc/media/engine/fakewebrtccall.h
+++ b/webrtc/media/engine/fakewebrtccall.h
@@ -35,8 +35,8 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
  public:
   struct TelephoneEvent {
     int payload_type = -1;
-    uint8_t event_code = 0;
-    uint32_t duration_ms = 0;
+    int event_code = 0;
+    int duration_ms = 0;
   };
 
   explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config);
@@ -56,8 +56,8 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
   }
 
   // webrtc::AudioSendStream implementation.
-  bool SendTelephoneEvent(int payload_type, uint8_t event,
-                          uint32_t duration_ms) override;
+  bool SendTelephoneEvent(int payload_type, int event,
+                          int duration_ms) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
   TelephoneEvent latest_telephone_event_;
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index f57db27a31..a496316b42 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1178,8 +1178,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     RTC_CHECK(stream_);
   }
 
-  bool SendTelephoneEvent(int payload_type, uint8_t event,
-                          uint32_t duration_ms) {
+  bool SendTelephoneEvent(int payload_type, int event, int duration_ms) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(stream_);
     return stream_->SendTelephoneEvent(payload_type, event, duration_ms);
diff --git a/webrtc/modules/audio_coding/test/TwoWayCommunication.cc b/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
index 3ca7fd217d..161491b061 100644
--- a/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
+++ b/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
@@ -50,14 +50,6 @@ TwoWayCommunication::~TwoWayCommunication() {
   delete _channel_B2A;
   delete _channelRef_A2B;
   delete _channelRef_B2A;
-#ifdef WEBRTC_DTMF_DETECTION
-  if (_dtmfDetectorA != NULL) {
-    delete _dtmfDetectorA;
-  }
-  if (_dtmfDetectorB != NULL) {
-    delete _dtmfDetectorB;
-  }
-#endif
   _inFileA.Close();
   _inFileB.Close();
   _outFileA.Close();
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 634969b311..32ab9379ae 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -322,7 +322,7 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
         (voice_codec.rate < 0) ? 0 : voice_codec.rate));
 
   // Start DTMF test.
-  uint32_t timeStamp = 160;
+  int timeStamp = 160;
 
   // Send a DTMF tone using RFC 2833 (4733).
   for (int i = 0; i < 16; i++) {
diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h
index f5c87334cc..c2211f8554 100644
--- a/webrtc/test/mock_voe_channel_proxy.h
+++ b/webrtc/test/mock_voe_channel_proxy.h
@@ -43,8 +43,7 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
   MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int32_t());
   MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
   MOCK_METHOD1(SetSendTelephoneEventPayloadType, bool(int payload_type));
-  MOCK_METHOD2(SendTelephoneEventOutband, bool(uint8_t event,
-                                               uint32_t duration_ms));
+  MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
 };
 }  // namespace test
 }  // namespace webrtc
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
index 9ccc53bf45..899b7333b2 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -18,8 +18,6 @@ source_set("voice_engine") {
     "channel_proxy.h",
     "dtmf_inband.cc",
     "dtmf_inband.h",
-    "dtmf_inband_queue.cc",
-    "dtmf_inband_queue.h",
     "include/voe_audio_processing.h",
     "include/voe_base.h",
     "include/voe_codec.h",
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 11af45edc8..695aa1dee2 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -40,13 +40,11 @@
 #include "webrtc/voice_engine/transmit_mixer.h"
 #include "webrtc/voice_engine/utility.h"
 
-#if defined(_WIN32)
-#include <Qos.h>
-#endif
-
 namespace webrtc {
 namespace voe {
 
+const int kTelephoneEventAttenuationdB = 10;
+
 class TransportFeedbackProxy : public TransportFeedbackObserver {
  public:
   TransportFeedbackProxy() : feedback_observer_(nullptr) {
@@ -363,7 +361,7 @@ void Channel::OnPlayTelephoneEvent(uint8_t event,
                " volume=%u)",
                event, lengthMs, volume);
 
-  if (!_playOutbandDtmfEvent || (event > 15)) {
+  if (event > 15) {
     // Ignore callback since feedback is disabled or event is not a
     // Dtmf tone event.
     return;
@@ -761,14 +759,11 @@ Channel::Channel(int32_t channelId,
       _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
       _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
       _outputFileRecording(false),
-      _inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
-      _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
       _outputExternalMedia(false),
       _inputExternalMediaCallbackPtr(NULL),
       _outputExternalMediaCallbackPtr(NULL),
       _timeStamp(0),  // This is just an offset, RTP module will add it's own
                       // random offset
-      _sendTelephoneEventPayloadType(106),
       ntp_estimator_(Clock::GetRealTimeClock()),
       jitter_buffer_playout_timestamp_(0),
       playout_timestamp_rtp_(0),
@@ -796,8 +791,6 @@ Channel::Channel(int32_t channelId,
       _panLeft(1.0f),
       _panRight(1.0f),
       _outputGain(1.0f),
-      _playOutbandDtmfEvent(false),
-      _playInbandDtmfEvent(false),
       _lastLocalTimeStamp(0),
       _lastPayloadType(0),
       _includeAudioLevelIndication(false),
@@ -830,8 +823,6 @@ Channel::Channel(int32_t channelId,
       config.Get<NetEqFastAccelerate>().enabled;
   audio_coding_.reset(AudioCodingModule::Create(acm_config));
 
-  _inbandDtmfQueue.ResetDtmf();
-  _inbandDtmfGenerator.Init();
   _outputAudioLevel.Clear();
 
   RtpRtcp::Configuration configuration;
@@ -2212,21 +2203,18 @@ int Channel::GetChannelOutputVolumeScaling(float& scaling) const {
   return 0;
 }
 
-int Channel::SendTelephoneEventOutband(unsigned char eventCode,
-                                       int lengthMs,
-                                       int attenuationDb,
-                                       bool playDtmfEvent) {
+int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::SendTelephoneEventOutband(..., playDtmfEvent=%d)",
-               playDtmfEvent);
+               "Channel::SendTelephoneEventOutband(...)");
+  RTC_DCHECK_LE(0, event);
+  RTC_DCHECK_GE(255, event);
+  RTC_DCHECK_LE(0, duration_ms);
+  RTC_DCHECK_GE(65535, duration_ms);
   if (!Sending()) {
     return -1;
   }
-
-  _playOutbandDtmfEvent = playDtmfEvent;
-
-  if (_rtpRtcpModule->SendTelephoneEventOutband(eventCode, lengthMs,
-                                                attenuationDb) != 0) {
+  if (_rtpRtcpModule->SendTelephoneEventOutband(
+      event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
     _engineStatisticsPtr->SetLastError(
         VE_SEND_DTMF_FAILED, kTraceWarning,
         "SendTelephoneEventOutband() failed to send event");
@@ -2235,32 +2223,14 @@ int Channel::SendTelephoneEventOutband(unsigned char eventCode,
   return 0;
 }
 
-int Channel::SendTelephoneEventInband(unsigned char eventCode,
-                                      int lengthMs,
-                                      int attenuationDb,
-                                      bool playDtmfEvent) {
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::SendTelephoneEventInband(..., playDtmfEvent=%d)",
-               playDtmfEvent);
-
-  _playInbandDtmfEvent = playDtmfEvent;
-  _inbandDtmfQueue.AddDtmf(eventCode, lengthMs, attenuationDb);
-
-  return 0;
-}
-
-int Channel::SetSendTelephoneEventPayloadType(unsigned char type) {
+int Channel::SetSendTelephoneEventPayloadType(int payload_type) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetSendTelephoneEventPayloadType()");
-  if (type > 127) {
-    _engineStatisticsPtr->SetLastError(
-        VE_INVALID_ARGUMENT, kTraceError,
-        "SetSendTelephoneEventPayloadType() invalid type");
-    return -1;
-  }
-  CodecInst codec = {};
+  RTC_DCHECK_LE(0, payload_type);
+  RTC_DCHECK_GE(127, payload_type);
+  CodecInst codec = {0};
   codec.plfreq = 8000;
-  codec.pltype = type;
+  codec.pltype = payload_type;
   memcpy(codec.plname, "telephone-event", 16);
   if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
     _rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
@@ -2272,12 +2242,6 @@ int Channel::SetSendTelephoneEventPayloadType(unsigned char type) {
       return -1;
     }
   }
-  _sendTelephoneEventPayloadType = type;
-  return 0;
-}
-
-int Channel::GetSendTelephoneEventPayloadType(unsigned char& type) {
-  type = _sendTelephoneEventPayloadType;
   return 0;
 }
 
@@ -3027,8 +2991,6 @@ uint32_t Channel::PrepareEncodeAndSend(int mixingFrequency) {
     }
   }
 
-  InsertInbandDtmfTone();
-
   if (_includeAudioLevelIndication) {
     size_t length =
         _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
@@ -3348,64 +3310,6 @@ int32_t Channel::MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency) {
   return 0;
 }
 
-int Channel::InsertInbandDtmfTone() {
-  // Check if we should start a new tone.
-  if (_inbandDtmfQueue.PendingDtmf() && !_inbandDtmfGenerator.IsAddingTone() &&
-      _inbandDtmfGenerator.DelaySinceLastTone() >
-          kMinTelephoneEventSeparationMs) {
-    int8_t eventCode(0);
-    uint16_t lengthMs(0);
-    uint8_t attenuationDb(0);
-
-    eventCode = _inbandDtmfQueue.NextDtmf(&lengthMs, &attenuationDb);
-    _inbandDtmfGenerator.AddTone(eventCode, lengthMs, attenuationDb);
-    if (_playInbandDtmfEvent) {
-      // Add tone to output mixer using a reduced length to minimize
-      // risk of echo.
-      _outputMixerPtr->PlayDtmfTone(eventCode, lengthMs - 80, attenuationDb);
-    }
-  }
-
-  if (_inbandDtmfGenerator.IsAddingTone()) {
-    uint16_t frequency(0);
-    _inbandDtmfGenerator.GetSampleRate(frequency);
-
-    if (frequency != _audioFrame.sample_rate_hz_) {
-      // Update sample rate of Dtmf tone since the mixing frequency
-      // has changed.
-      _inbandDtmfGenerator.SetSampleRate(
-          (uint16_t)(_audioFrame.sample_rate_hz_));
-      // Reset the tone to be added taking the new sample rate into
-      // account.
-      _inbandDtmfGenerator.ResetTone();
-    }
-
-    int16_t toneBuffer[320];
-    uint16_t toneSamples(0);
-    // Get 10ms tone segment and set time since last tone to zero
-    if (_inbandDtmfGenerator.Get10msTone(toneBuffer, toneSamples) == -1) {
-      WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
-                   "Channel::EncodeAndSend() inserting Dtmf failed");
-      return -1;
-    }
-
-    // Replace mixed audio with DTMF tone.
-    for (size_t sample = 0; sample < _audioFrame.samples_per_channel_;
-         sample++) {
-      for (size_t channel = 0; channel < _audioFrame.num_channels_; channel++) {
-        const size_t index = sample * _audioFrame.num_channels_ + channel;
-        _audioFrame.data_[index] = toneBuffer[sample];
-      }
-    }
-
-    assert(_audioFrame.samples_per_channel_ == toneSamples);
-  } else {
-    // Add 10ms to "delay-since-last-tone" counter
-    _inbandDtmfGenerator.UpdateDelaySinceLastTone();
-  }
-  return 0;
-}
-
 void Channel::UpdatePlayoutTimestamp(bool rtcp) {
   uint32_t playout_timestamp = 0;
 
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 75c4fd87cb..c89b0e0c8b 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -25,8 +25,6 @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/file_player.h"
 #include "webrtc/modules/utility/include/file_recorder.h"
-#include "webrtc/voice_engine/dtmf_inband.h"
-#include "webrtc/voice_engine/dtmf_inband_queue.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/level_indicator.h"
@@ -296,17 +294,9 @@ class Channel
   // VoEVideoSyncExtended
   int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
 
-  // VoEDtmf
-  int SendTelephoneEventOutband(unsigned char eventCode,
-                                int lengthMs,
-                                int attenuationDb,
-                                bool playDtmfEvent);
-  int SendTelephoneEventInband(unsigned char eventCode,
-                               int lengthMs,
-                               int attenuationDb,
-                               bool playDtmfEvent);
-  int SetSendTelephoneEventPayloadType(unsigned char type);
-  int GetSendTelephoneEventPayloadType(unsigned char& type);
+  // DTMF
+  int SendTelephoneEventOutband(int event, int duration_ms);
+  int SetSendTelephoneEventPayloadType(int payload_type);
 
   // VoEAudioProcessingImpl
   int UpdateRxVadDetection(AudioFrame& audioFrame);
@@ -464,7 +454,6 @@ class Channel
   bool IsPacketInOrder(const RTPHeader& header) const;
   bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
   int ResendPackets(const uint16_t* sequence_numbers, int length);
-  int InsertInbandDtmfTone();
   int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
   int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
   void UpdatePlayoutTimestamp(bool rtcp);
@@ -510,13 +499,10 @@ class Channel
   int _outputFilePlayerId;
   int _outputFileRecorderId;
   bool _outputFileRecording;
-  DtmfInbandQueue _inbandDtmfQueue;
-  DtmfInband _inbandDtmfGenerator;
   bool _outputExternalMedia;
   VoEMediaProcess* _inputExternalMediaCallbackPtr;
   VoEMediaProcess* _outputExternalMediaCallbackPtr;
   uint32_t _timeStamp;
-  uint8_t _sendTelephoneEventPayloadType;
 
   RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
 
@@ -560,9 +546,6 @@ class Channel
   float _panLeft;
   float _panRight;
   float _outputGain;
-  // VoEDtmf
-  bool _playOutbandDtmfEvent;
-  bool _playInbandDtmfEvent;
   // VoeRTP_RTCP
   uint32_t _lastLocalTimeStamp;
   int8_t _lastPayloadType;
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index da7864f15f..10c8821202 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -148,11 +148,9 @@ bool ChannelProxy::SetSendTelephoneEventPayloadType(int payload_type) {
   return channel()->SetSendTelephoneEventPayloadType(payload_type) == 0;
 }
 
-bool ChannelProxy::SendTelephoneEventOutband(uint8_t event,
-                                             uint32_t duration_ms) {
+bool ChannelProxy::SendTelephoneEventOutband(int event, int duration_ms) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  return
-      channel()->SendTelephoneEventOutband(event, duration_ms, 10, false) == 0;
+  return channel()->SendTelephoneEventOutband(event, duration_ms) == 0;
 }
 
 void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index 3461cf3e78..dec726cb23 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -68,7 +68,7 @@ class ChannelProxy {
   virtual uint32_t GetDelayEstimate() const;
 
   virtual bool SetSendTelephoneEventPayloadType(int payload_type);
-  virtual bool SendTelephoneEventOutband(uint8_t event, uint32_t duration_ms);
+  virtual bool SendTelephoneEventOutband(int event, int duration_ms);
 
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
 
diff --git a/webrtc/voice_engine/dtmf_inband.h b/webrtc/voice_engine/dtmf_inband.h
index 6000d99214..795c5ce8fc 100644
--- a/webrtc/voice_engine/dtmf_inband.h
+++ b/webrtc/voice_engine/dtmf_inband.h
@@ -17,6 +17,9 @@
 
 namespace webrtc {
 
+// TODO(solenberg): Used as a DTMF tone generator in voe::OutputMixer. Pull out
+//                  the one in NetEq and use that instead? We don't need several
+//                  implemenations of this.
 class DtmfInband
 {
 public:
diff --git a/webrtc/voice_engine/dtmf_inband_queue.cc b/webrtc/voice_engine/dtmf_inband_queue.cc
deleted file mode 100644
index 4ab74cdf70..0000000000
--- a/webrtc/voice_engine/dtmf_inband_queue.cc
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/voice_engine/dtmf_inband_queue.h"
-
-namespace webrtc {
-
-DtmfInbandQueue::DtmfInbandQueue(int32_t id):
-    _id(id),
-    _nextEmptyIndex(0)
-{
-    memset(_DtmfKey,0, sizeof(_DtmfKey));
-    memset(_DtmfLen,0, sizeof(_DtmfLen));
-    memset(_DtmfLevel,0, sizeof(_DtmfLevel));
-}
-
-DtmfInbandQueue::~DtmfInbandQueue()
-{
-}
-
-int
-DtmfInbandQueue::AddDtmf(uint8_t key, uint16_t len, uint8_t level)
-{
-    rtc::CritScope lock(&_DtmfCritsect);
-
-    if (_nextEmptyIndex >= kDtmfInbandMax)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_id,-1),
-                   "DtmfInbandQueue::AddDtmf() unable to add Dtmf tone");
-        return -1;
-    }
-    int32_t index = _nextEmptyIndex;
-    _DtmfKey[index] = key;
-    _DtmfLen[index] = len;
-    _DtmfLevel[index] = level;
-    _nextEmptyIndex++;
-    return 0;
-}
-
-int8_t
-DtmfInbandQueue::NextDtmf(uint16_t* len, uint8_t* level)
-{
-    rtc::CritScope lock(&_DtmfCritsect);
-
-    if(!PendingDtmf())
-    {
-        return -1;
-    }
-    int8_t nextDtmf = _DtmfKey[0];
-    *len=_DtmfLen[0];
-    *level=_DtmfLevel[0];
-
-    memmove(&(_DtmfKey[0]), &(_DtmfKey[1]),
-            _nextEmptyIndex*sizeof(uint8_t));
-    memmove(&(_DtmfLen[0]), &(_DtmfLen[1]),
-            _nextEmptyIndex*sizeof(uint16_t));
-    memmove(&(_DtmfLevel[0]), &(_DtmfLevel[1]),
-            _nextEmptyIndex*sizeof(uint8_t));
-
-    _nextEmptyIndex--;
-    return nextDtmf;
-}
-
-bool
-DtmfInbandQueue::PendingDtmf()
-{
-    rtc::CritScope lock(&_DtmfCritsect);
-    return _nextEmptyIndex > 0;
-}
-
-void
-DtmfInbandQueue::ResetDtmf()
-{
-    rtc::CritScope lock(&_DtmfCritsect);
-    _nextEmptyIndex = 0;
-}
-
-}  // namespace webrtc
diff --git a/webrtc/voice_engine/dtmf_inband_queue.h b/webrtc/voice_engine/dtmf_inband_queue.h
deleted file mode 100644
index 08e8018316..0000000000
--- a/webrtc/voice_engine/dtmf_inband_queue.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
-#define WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
-
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/voice_engine/voice_engine_defines.h"
-
-
-namespace webrtc {
-
-class DtmfInbandQueue
-{
-public:
-
-    DtmfInbandQueue(int32_t id);
-
-    virtual ~DtmfInbandQueue();
-
-    int AddDtmf(uint8_t DtmfKey, uint16_t len, uint8_t level);
-
-    int8_t NextDtmf(uint16_t* len, uint8_t* level);
-
-    bool PendingDtmf();
-
-    void ResetDtmf();
-
-private:
-    enum {kDtmfInbandMax = 20};
-
-    int32_t _id;
-    rtc::CriticalSection _DtmfCritsect;
-    uint8_t _nextEmptyIndex;
-    uint8_t _DtmfKey[kDtmfInbandMax];
-    uint16_t _DtmfLen[kDtmfInbandMax];
-    uint8_t _DtmfLevel[kDtmfInbandMax];
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index d6a5213217..0240389be1 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -205,7 +205,6 @@ TransmitMixer::TransmitMixer(uint32_t instanceId) :
     external_postproc_ptr_(NULL),
     external_preproc_ptr_(NULL),
     _mute(false),
-    _remainingMuteMicTimeMs(0),
     stereo_codec_(false),
     swap_stereo_channels_(false)
 {
@@ -359,17 +358,6 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
     TypingDetection(keyPressed);
 #endif
 
-    // --- Mute during DTMF tone if direct feedback is enabled
-    if (_remainingMuteMicTimeMs > 0)
-    {
-        AudioFrameOperations::Mute(_audioFrame);
-        _remainingMuteMicTimeMs -= 10;
-        if (_remainingMuteMicTimeMs < 0)
-        {
-            _remainingMuteMicTimeMs = 0;
-        }
-    }
-
     // --- Mute signal
     if (_mute)
     {
@@ -477,15 +465,6 @@ uint32_t TransmitMixer::CaptureLevel() const
     return _captureLevel;
 }
 
-void
-TransmitMixer::UpdateMuteMicrophoneTime(uint32_t lengthMs)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
-               "TransmitMixer::UpdateMuteMicrophoneTime(lengthMs=%d)",
-               lengthMs);
-    _remainingMuteMicTimeMs = lengthMs;
-}
-
 int32_t
 TransmitMixer::StopSend()
 {
diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h
index b5c483aa4b..483af0518a 100644
--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/voice_engine/transmit_mixer.h
@@ -75,9 +75,6 @@ public:
 
     int32_t StopSend();
 
-    // VoEDtmf
-    void UpdateMuteMicrophoneTime(uint32_t lengthMs);
-
     // VoEExternalMedia
     int RegisterExternalMediaProcessing(VoEMediaProcess* object,
                                         ProcessingTypes type);
@@ -226,7 +223,6 @@ private:
     VoEMediaProcess* external_postproc_ptr_;
     VoEMediaProcess* external_preproc_ptr_;
     bool _mute;
-    int32_t _remainingMuteMicTimeMs;
     bool stereo_codec_;
     bool swap_stereo_channels_;
 };
diff --git a/webrtc/voice_engine/voice_engine.gyp b/webrtc/voice_engine/voice_engine.gyp
index 7935be89cf..a848d6c668 100644
--- a/webrtc/voice_engine/voice_engine.gyp
+++ b/webrtc/voice_engine/voice_engine.gyp
@@ -54,8 +54,6 @@
         'channel_proxy.h',
         'dtmf_inband.cc',
         'dtmf_inband.h',
-        'dtmf_inband_queue.cc',
-        'dtmf_inband_queue.h',
         'level_indicator.cc',
         'level_indicator.h',
         'monitor_module.cc',