diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 680652322..d91cbd2fd 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -12,7 +12,6 @@
 
 #include <assert.h>
 
-#include "webrtc/base/fileutils.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
@@ -717,11 +716,6 @@ int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
-int AudioProcessingImpl::StartDebugRecording(rtc::PlatformFile handle) {
-  FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
-  return StartDebugRecording(stream);
-}
-
 int AudioProcessingImpl::StopDebugRecording() {
   CriticalSectionScoped crit_scoped(crit_);
 
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index d012e7f0e..9753423d6 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -125,7 +125,6 @@ class AudioProcessingImpl : public AudioProcessing {
   virtual int StartDebugRecording(
       const char filename[kMaxFilenameSize]) OVERRIDE;
   virtual int StartDebugRecording(FILE* handle) OVERRIDE;
-  virtual int StartDebugRecording(rtc::PlatformFile handle) OVERRIDE;
   virtual int StopDebugRecording() OVERRIDE;
   virtual EchoCancellation* echo_cancellation() const OVERRIDE;
   virtual EchoControlMobile* echo_control_mobile() const OVERRIDE;
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index 53157ae66..30f0d9c5d 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -14,7 +14,6 @@
 #include <stddef.h>  // size_t
 #include <stdio.h>  // FILE
 
-#include "webrtc/base/fileutils.h"
 #include "webrtc/common.h"
 #include "webrtc/typedefs.h"
 
@@ -326,11 +325,6 @@ class AudioProcessing {
   // of |handle| and closes it at StopDebugRecording().
   virtual int StartDebugRecording(FILE* handle) = 0;
 
-  // Same as above but uses an existing PlatformFile handle. Takes ownership
-  // of |handle| and closes it at StopDebugRecording().
-  // TODO(xians): Make this interface pure virtual.
-  virtual int StartDebugRecording(rtc::PlatformFile handle) { return -1; }
-
   // Stops recording debugging information, and closes the file. Recording
   // cannot be resumed in the same file (without overwriting it).
   virtual int StopDebugRecording() = 0;