diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 3ea59a80c1..bc95839392 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -187,7 +187,7 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEAudioProcessing,
       public webrtc::VoEBase, public webrtc::VoECodec, public webrtc::VoEDtmf,
-      public webrtc::VoEFile, public webrtc::VoEHardware,
+      public webrtc::VoEHardware,
       public webrtc::VoEExternalMedia, public webrtc::VoENetEqStats,
       public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
       public webrtc::VoEVideoSync, public webrtc::VoEVolumeControl {
@@ -209,7 +209,6 @@ class FakeWebRtcVoiceEngine
           volume_scale(1.0),
           volume_pan_left(1.0),
           volume_pan_right(1.0),
-          file(false),
           vad(false),
           codec_fec(false),
           max_encoding_bandwidth(0),
@@ -241,7 +240,6 @@ class FakeWebRtcVoiceEngine
     float volume_scale;
     float volume_pan_left;
     float volume_pan_right;
-    bool file;
     bool vad;
     bool codec_fec;
     int max_encoding_bandwidth;
@@ -723,64 +721,6 @@ class FakeWebRtcVoiceEngine
     return 0;
   }
 
-  // webrtc::VoEFile
-  WEBRTC_FUNC(StartPlayingFileLocally, (int channel, const char* fileNameUTF8,
-                                        bool loop, webrtc::FileFormats format,
-                                        float volumeScaling, int startPointMs,
-                                        int stopPointMs)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->file = true;
-    return 0;
-  }
-  WEBRTC_FUNC(StartPlayingFileLocally, (int channel, webrtc::InStream* stream,
-                                        webrtc::FileFormats format,
-                                        float volumeScaling, int startPointMs,
-                                        int stopPointMs)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->file = true;
-    return 0;
-  }
-  WEBRTC_FUNC(StopPlayingFileLocally, (int channel)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->file = false;
-    return 0;
-  }
-  WEBRTC_FUNC(IsPlayingFileLocally, (int channel)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    return (channels_[channel]->file) ? 1 : 0;
-  }
-  WEBRTC_STUB(StartPlayingFileAsMicrophone, (int channel,
-                                             const char* fileNameUTF8,
-                                             bool loop,
-                                             bool mixWithMicrophone,
-                                             webrtc::FileFormats format,
-                                             float volumeScaling));
-  WEBRTC_STUB(StartPlayingFileAsMicrophone, (int channel,
-                                             webrtc::InStream* stream,
-                                             bool mixWithMicrophone,
-                                             webrtc::FileFormats format,
-                                             float volumeScaling));
-  WEBRTC_STUB(StopPlayingFileAsMicrophone, (int channel));
-  WEBRTC_STUB(IsPlayingFileAsMicrophone, (int channel));
-  WEBRTC_STUB(StartRecordingPlayout, (int channel, const char* fileNameUTF8,
-                                      webrtc::CodecInst* compression,
-                                      int maxSizeBytes));
-  WEBRTC_STUB(StartRecordingPlayout, (int channel, webrtc::OutStream* stream,
-                                      webrtc::CodecInst* compression));
-  WEBRTC_STUB(StopRecordingPlayout, (int channel));
-  WEBRTC_FUNC(StartRecordingMicrophone, (const char* fileNameUTF8,
-                                         webrtc::CodecInst* compression,
-                                         int maxSizeBytes)) {
-    return 0;
-  }
-  WEBRTC_FUNC(StartRecordingMicrophone, (webrtc::OutStream* stream,
-                                         webrtc::CodecInst* compression)) {
-    return 0;
-  }
-  WEBRTC_FUNC(StopRecordingMicrophone, ()) {
-    return 0;
-  }
-
   // webrtc::VoEHardware
   WEBRTC_FUNC(GetNumOfRecordingDevices, (int& num)) {
     return GetNumDevices(num);
diff --git a/talk/media/webrtc/webrtcvoe.h b/talk/media/webrtc/webrtcvoe.h
index 40e955ad1e..dad2f3989f 100644
--- a/talk/media/webrtc/webrtcvoe.h
+++ b/talk/media/webrtc/webrtcvoe.h
@@ -95,7 +95,7 @@ class VoEWrapper {
  public:
   VoEWrapper()
       : engine_(webrtc::VoiceEngine::Create()), processing_(engine_),
-        base_(engine_), codec_(engine_), dtmf_(engine_), file_(engine_),
+        base_(engine_), codec_(engine_), dtmf_(engine_),
         hw_(engine_), media_(engine_), neteq_(engine_), network_(engine_),
         rtp_(engine_), sync_(engine_), volume_(engine_) {
   }
@@ -103,7 +103,6 @@ class VoEWrapper {
              webrtc::VoEBase* base,
              webrtc::VoECodec* codec,
              webrtc::VoEDtmf* dtmf,
-             webrtc::VoEFile* file,
              webrtc::VoEHardware* hw,
              webrtc::VoEExternalMedia* media,
              webrtc::VoENetEqStats* neteq,
@@ -116,7 +115,6 @@ class VoEWrapper {
         base_(base),
         codec_(codec),
         dtmf_(dtmf),
-        file_(file),
         hw_(hw),
         media_(media),
         neteq_(neteq),
@@ -131,7 +129,6 @@ class VoEWrapper {
   webrtc::VoEBase* base() const { return base_.get(); }
   webrtc::VoECodec* codec() const { return codec_.get(); }
   webrtc::VoEDtmf* dtmf() const { return dtmf_.get(); }
-  webrtc::VoEFile* file() const { return file_.get(); }
   webrtc::VoEHardware* hw() const { return hw_.get(); }
   webrtc::VoEExternalMedia* media() const { return media_.get(); }
   webrtc::VoENetEqStats* neteq() const { return neteq_.get(); }
@@ -147,7 +144,6 @@ class VoEWrapper {
   scoped_voe_ptr<webrtc::VoEBase> base_;
   scoped_voe_ptr<webrtc::VoECodec> codec_;
   scoped_voe_ptr<webrtc::VoEDtmf> dtmf_;
-  scoped_voe_ptr<webrtc::VoEFile> file_;
   scoped_voe_ptr<webrtc::VoEHardware> hw_;
   scoped_voe_ptr<webrtc::VoEExternalMedia> media_;
   scoped_voe_ptr<webrtc::VoENetEqStats> neteq_;
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 358645d853..6157b22c71 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -2308,11 +2308,6 @@ bool WebRtcVoiceMediaChannel::ChangeSend(int channel, SendFlags send) {
       LOG_RTCERR1(StartSend, channel);
       return false;
     }
-    if (engine()->voe()->file() &&
-        engine()->voe()->file()->StopPlayingFileAsMicrophone(channel) == -1) {
-      LOG_RTCERR1(StopPlayingFileAsMicrophone, channel);
-      return false;
-    }
   } else {  // SEND_NOTHING
     RTC_DCHECK(send == SEND_NOTHING);
     if (engine()->voe()->base()->StopSend(channel) == -1) {
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 5a04fb2a4b..85c3dc7f9b 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -47,10 +47,6 @@
 #include "webrtc/common.h"
 #include "webrtc/config.h"
 
-namespace webrtc {
-class VideoEngine;
-}
-
 namespace cricket {
 
 class AudioDeviceModule;
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index d37374e972..75273f3812 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -68,7 +68,6 @@ class FakeVoEWrapper : public cricket::VoEWrapper {
                             engine,  // base
                             engine,  // codec
                             engine,  // dtmf
-                            engine,  // file
                             engine,  // hw
                             engine,  // media
                             engine,  // neteq