diff --git a/src/modules/media_file/source/media_file_impl.cc b/src/modules/media_file/source/media_file_impl.cc
index 3199c1246..869830f78 100644
--- a/src/modules/media_file/source/media_file_impl.cc
+++ b/src/modules/media_file/source/media_file_impl.cc
@@ -1150,9 +1150,9 @@ WebRtc_Word32 MediaFileImpl::StartRecordingStream(
             StopRecording();
             return -1;
         }
-        if((STR_NCASE_CMP(codec_info_.plname, "L16", 4) != 0) &&
-           (STR_NCASE_CMP(codec_info_.plname, "PCMU", 5) != 0) &&
-           (STR_NCASE_CMP(codec_info_.plname, "PCMA", 5) != 0))
+        if((STR_NCASE_CMP(tmpAudioCodec.plname, "L16", 4) != 0) &&
+           (STR_NCASE_CMP(tmpAudioCodec.plname, "PCMU", 5) != 0) &&
+           (STR_NCASE_CMP(tmpAudioCodec.plname, "PCMA", 5) != 0))
         {
             WEBRTC_TRACE(
                 kTraceWarning,
diff --git a/src/modules/utility/source/file_recorder_impl.cc b/src/modules/utility/source/file_recorder_impl.cc
index ecba43579..7fff4b4a9 100644
--- a/src/modules/utility/source/file_recorder_impl.cc
+++ b/src/modules/utility/source/file_recorder_impl.cc
@@ -204,8 +204,10 @@ WebRtc_Word32 FileRecorderImpl::RecordAudioToFile(
         // Recording mono but incoming audio is (interleaved) stereo.
         tempAudioFrame._audioChannel = 1;
         tempAudioFrame._frequencyInHz = incomingAudioFrame._frequencyInHz;
+        tempAudioFrame._payloadDataLengthInSamples =
+          incomingAudioFrame._payloadDataLengthInSamples;
         for (WebRtc_UWord16 i = 0;
-             i < (incomingAudioFrame._payloadDataLengthInSamples >> 1); i++)
+             i < (incomingAudioFrame._payloadDataLengthInSamples); i++)
         {
             // Sample value is the average of left and right buffer rounded to
             // closest integer value. Note samples can be either 1 or 2 byte.
@@ -213,8 +215,24 @@ WebRtc_Word32 FileRecorderImpl::RecordAudioToFile(
                  ((incomingAudioFrame._payloadData[2 * i] +
                    incomingAudioFrame._payloadData[(2 * i) + 1] + 1) >> 1);
         }
+    }
+    else if( incomingAudioFrame._audioChannel == 1 &&
+        _moduleFile->IsStereo())
+    {
+        // Recording stereo but incoming audio is mono.
+        tempAudioFrame._audioChannel = 2;
+        tempAudioFrame._frequencyInHz = incomingAudioFrame._frequencyInHz;
         tempAudioFrame._payloadDataLengthInSamples =
-            incomingAudioFrame._payloadDataLengthInSamples / 2;
+          incomingAudioFrame._payloadDataLengthInSamples;
+        for (WebRtc_UWord16 i = 0;
+             i < (incomingAudioFrame._payloadDataLengthInSamples); i++)
+        {
+            // Duplicate sample to both channels
+             tempAudioFrame._payloadData[2*i] =
+               incomingAudioFrame._payloadData[i];
+             tempAudioFrame._payloadData[2*i+1] =
+               incomingAudioFrame._payloadData[i];
+        }
     }
 
     const AudioFrame* ptrAudioFrame = &incomingAudioFrame;
@@ -254,7 +272,8 @@ WebRtc_Word32 FileRecorderImpl::RecordAudioToFile(
                                           codec_info_.plfreq,
                                           kResamplerSynchronousStereo);
             _audioResampler.Push(ptrAudioFrame->_payloadData,
-                                 ptrAudioFrame->_payloadDataLengthInSamples,
+                                 ptrAudioFrame->_payloadDataLengthInSamples *
+                                 ptrAudioFrame->_audioChannel,
                                  (WebRtc_Word16*)_audioBuffer,
                                  MAX_AUDIO_BUFFER_IN_BYTES, outLen);
         } else {
@@ -266,7 +285,7 @@ WebRtc_Word32 FileRecorderImpl::RecordAudioToFile(
                                  (WebRtc_Word16*)_audioBuffer,
                                  MAX_AUDIO_BUFFER_IN_BYTES, outLen);
         }
-        encodedLenInBytes = outLen*2;
+        encodedLenInBytes = outLen * sizeof(WebRtc_Word16);
     }
 
     // Codec may not be operating at a frame rate of 10 ms. Whenever enough
diff --git a/src/voice_engine/main/source/transmit_mixer.cc b/src/voice_engine/main/source/transmit_mixer.cc
index 25f70d37c..d131a8574 100644
--- a/src/voice_engine/main/source/transmit_mixer.cc
+++ b/src/voice_engine/main/source/transmit_mixer.cc
@@ -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
@@ -722,7 +722,8 @@ int TransmitMixer::StartRecordingMicrophone(const WebRtc_Word8* fileName,
     const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
     CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
 
-    if (codecInst != NULL && codecInst->channels != 1)
+    if (codecInst != NULL &&
+      (codecInst->channels < 0 || codecInst->channels > 2))
     {
         _engineStatisticsPtr->SetLastError(
             VE_BAD_ARGUMENT, kTraceError,
@@ -1194,8 +1195,6 @@ TransmitMixer::GenerateAudioFrame(const WebRtc_Word16 audioSamples[],
 WebRtc_Word32 TransmitMixer::RecordAudioToFile(
     const WebRtc_UWord32 mixingFrequency)
 {
-    assert(_audioFrame._audioChannel == 1);
-
     CriticalSectionScoped cs(_critSect);
     if (_fileRecorderPtr == NULL)
     {