diff --git a/webrtc/modules/audio_processing/audio_processing.gypi b/webrtc/modules/audio_processing/audio_processing.gypi
index 9bbcfae7f8..969fdb12c3 100644
--- a/webrtc/modules/audio_processing/audio_processing.gypi
+++ b/webrtc/modules/audio_processing/audio_processing.gypi
@@ -9,6 +9,7 @@
 {
   'variables': {
     'audio_processing_dependencies': [
+      '<(webrtc_root)/base/base.gyp:rtc_base',
       '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
       '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
     ],
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index d91cbd2fd3..6806523221 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -12,6 +12,7 @@
 
 #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"
@@ -716,6 +717,11 @@ 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 9753423d6d..d012e7f0ea 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -125,6 +125,7 @@ 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 30f0d9c5d9..53157ae66f 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -14,6 +14,7 @@
 #include <stddef.h>  // size_t
 #include <stdio.h>  // FILE
 
+#include "webrtc/base/fileutils.h"
 #include "webrtc/common.h"
 #include "webrtc/typedefs.h"
 
@@ -325,6 +326,11 @@ 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;