diff --git a/src/modules/audio_coding/main/source/audio_coding_module_impl.cc b/src/modules/audio_coding/main/source/audio_coding_module_impl.cc
index 2abe46b7e..38c973db6 100644
--- a/src/modules/audio_coding/main/source/audio_coding_module_impl.cc
+++ b/src/modules/audio_coding/main/source/audio_coding_module_impl.cc
@@ -2002,25 +2002,24 @@ AudioCodingModuleImpl::PlayoutData10Ms(
     AudioFrame&         audioFrame)
 {
     bool stereoMode;
-    AudioFrame audioFrameTmp;
 
      // recOut always returns 10 ms
-    if (_netEq.RecOut(audioFrameTmp) != 0)
+    if (_netEq.RecOut(_audioFrame) != 0)
     {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
             "PlayoutData failed, RecOut Failed");
         return -1;
     }
 
-    audioFrame._audioChannel = audioFrameTmp._audioChannel;
-    audioFrame._vadActivity  = audioFrameTmp._vadActivity;
-    audioFrame._speechType   = audioFrameTmp._speechType;
+    audioFrame._audioChannel = _audioFrame._audioChannel;
+    audioFrame._vadActivity  = _audioFrame._vadActivity;
+    audioFrame._speechType   = _audioFrame._speechType;
 
-    stereoMode =  (audioFrameTmp._audioChannel > 1);
+    stereoMode =  (_audioFrame._audioChannel > 1);
     //For stereo playout:
     // Master and Slave samples are interleaved starting with Master
 
-    const WebRtc_UWord16 recvFreq = static_cast<WebRtc_UWord16>(audioFrameTmp._frequencyInHz);
+    const WebRtc_UWord16 recvFreq = static_cast<WebRtc_UWord16>(_audioFrame._frequencyInHz);
     bool toneDetected = false;
     WebRtc_Word16 lastDetectedTone;
     WebRtc_Word16 tone;
@@ -2036,8 +2035,8 @@ AudioCodingModuleImpl::PlayoutData10Ms(
         {
             // resample payloadData
             WebRtc_Word16 tmpLen = _outputResampler.Resample10Msec(
-                audioFrameTmp._payloadData, recvFreq, audioFrame._payloadData, desiredFreqHz,
-                audioFrameTmp._audioChannel);
+                _audioFrame._payloadData, recvFreq, audioFrame._payloadData, desiredFreqHz,
+                _audioFrame._audioChannel);
 
             if(tmpLen < 0)
             {
@@ -2053,11 +2052,11 @@ AudioCodingModuleImpl::PlayoutData10Ms(
         }
         else
         {
-            memcpy(audioFrame._payloadData, audioFrameTmp._payloadData,
-              audioFrameTmp._payloadDataLengthInSamples * audioFrame._audioChannel
+            memcpy(audioFrame._payloadData, _audioFrame._payloadData,
+              _audioFrame._payloadDataLengthInSamples * audioFrame._audioChannel
               * sizeof(WebRtc_Word16));
             // set the payload length
-            audioFrame._payloadDataLengthInSamples = audioFrameTmp._payloadDataLengthInSamples;
+            audioFrame._payloadDataLengthInSamples = _audioFrame._payloadDataLengthInSamples;
             // set the sampling frequency
             audioFrame._frequencyInHz = recvFreq;
         }
@@ -2091,22 +2090,22 @@ AudioCodingModuleImpl::PlayoutData10Ms(
             }
             else
             {
-                // Do the detection on the audio that we got from NetEQ (audioFrameTmp).
+                // Do the detection on the audio that we got from NetEQ (_audioFrame).
                 if(!stereoMode)
                 {
-                    _dtmfDetector->Detect(audioFrameTmp._payloadData,
-                        audioFrameTmp._payloadDataLengthInSamples, recvFreq,
+                    _dtmfDetector->Detect(_audioFrame._payloadData,
+                        _audioFrame._payloadDataLengthInSamples, recvFreq,
                         toneDetected, tone);
                 }
                 else
                 {
                     WebRtc_Word16 masterChannel[WEBRTC_10MS_PCM_AUDIO];
-                    for(int n = 0; n < audioFrameTmp._payloadDataLengthInSamples; n++)
+                    for(int n = 0; n < _audioFrame._payloadDataLengthInSamples; n++)
                     {
-                        masterChannel[n] = audioFrameTmp._payloadData[n<<1];
+                        masterChannel[n] = _audioFrame._payloadData[n<<1];
                     }
                     _dtmfDetector->Detect(masterChannel,
-                        audioFrameTmp._payloadDataLengthInSamples, recvFreq,
+                        _audioFrame._payloadDataLengthInSamples, recvFreq,
                         toneDetected, tone);
                 }
             }
diff --git a/src/modules/audio_coding/main/source/audio_coding_module_impl.h b/src/modules/audio_coding/main/source/audio_coding_module_impl.h
index d81d7fc82..365472865 100644
--- a/src/modules/audio_coding/main/source/audio_coding_module_impl.h
+++ b/src/modules/audio_coding/main/source/audio_coding_module_impl.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -370,6 +370,8 @@ private:
     TimedTrace                     _trace;
 #endif
 
+    AudioFrame                     _audioFrame;
+
 #ifdef ACM_QA_TEST
     FILE* _outgoingPL;
     FILE* _incomingPL;
diff --git a/src/voice_engine/main/source/voe_base_impl.cc b/src/voice_engine/main/source/voe_base_impl.cc
index 6fe6ba9af..e1cd08377 100644
--- a/src/voice_engine/main/source/voe_base_impl.cc
+++ b/src/voice_engine/main/source/voe_base_impl.cc
@@ -259,8 +259,6 @@ WebRtc_Word32 VoEBaseImpl::NeedMorePlayData(
 
     assert(_outputMixerPtr != NULL);
 
-    AudioFrame audioFrame;
-
     // Perform mixing of all active participants (channel-based mixing)
     _outputMixerPtr->MixActiveChannels();
 
@@ -268,20 +266,20 @@ WebRtc_Word32 VoEBaseImpl::NeedMorePlayData(
     _outputMixerPtr->DoOperationsOnCombinedSignal();
 
     // Retrieve the final output mix (resampled to match the ADM)
-    _outputMixerPtr->GetMixedAudio(samplesPerSec, nChannels, audioFrame);
+    _outputMixerPtr->GetMixedAudio(samplesPerSec, nChannels, _audioFrame);
 
-    assert(nSamples == audioFrame._payloadDataLengthInSamples);
+    assert(nSamples == _audioFrame._payloadDataLengthInSamples);
     assert(samplesPerSec ==
-        static_cast<WebRtc_UWord32>(audioFrame._frequencyInHz));
+        static_cast<WebRtc_UWord32>(_audioFrame._frequencyInHz));
 
     // Deliver audio (PCM) samples to the ADM
     memcpy(
            (WebRtc_Word16*) audioSamples,
-           (const WebRtc_Word16*) audioFrame._payloadData,
-           sizeof(WebRtc_Word16) * (audioFrame._payloadDataLengthInSamples
-                   * audioFrame._audioChannel));
+           (const WebRtc_Word16*) _audioFrame._payloadData,
+           sizeof(WebRtc_Word16) * (_audioFrame._payloadDataLengthInSamples
+                   * _audioFrame._audioChannel));
 
-    nSamplesOut = audioFrame._payloadDataLengthInSamples;
+    nSamplesOut = _audioFrame._payloadDataLengthInSamples;
 
     return 0;
 }
diff --git a/src/voice_engine/main/source/voe_base_impl.h b/src/voice_engine/main/source/voe_base_impl.h
index 922187db2..6d470a0ff 100644
--- a/src/voice_engine/main/source/voe_base_impl.h
+++ b/src/voice_engine/main/source/voe_base_impl.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -13,6 +13,7 @@
 
 #include "voe_base.h"
 
+#include "module_common_types.h"
 #include "ref_count.h"
 #include "shared_data.h"
 
@@ -142,6 +143,8 @@ private:
     bool _voiceEngineObserver;
     WebRtc_UWord32 _oldVoEMicLevel;
     WebRtc_UWord32 _oldMicLevel;
+    AudioFrame _audioFrame;
+
 };
 
 } // namespace webrtc