diff --git a/src/modules/audio_coding/main/source/acm_g7291.cc b/src/modules/audio_coding/main/source/acm_g7291.cc
index 043beb2c5..d832e9c7a 100644
--- a/src/modules/audio_coding/main/source/acm_g7291.cc
+++ b/src/modules/audio_coding/main/source/acm_g7291.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
@@ -44,7 +44,10 @@ namespace webrtc {
 
 ACMG729_1::ACMG729_1( WebRtc_Word16 /* codecID */)
     : _encoderInstPtr(NULL),
-      _decoderInstPtr(NULL) {
+      _decoderInstPtr(NULL),
+      _myRate(32000),
+      _flag8kHz(0),
+      _flagG729mode(0) {
   return;
 }
 
@@ -163,22 +166,16 @@ ACMG729_1::SetBitRateSafe(
 
 struct G729_1_inst_t_;
 
-ACMG729_1::ACMG729_1(
-    WebRtc_Word16 codecID):
-_encoderInstPtr(NULL),
-_decoderInstPtr(NULL)
-{
-    _codecID = codecID;
-    // Our current G729.1 does not support Annex C
-    // which is DTX.
-    _hasInternalDTX = false;
-
-  // Default rate
-    _myRate = 32000;
-
-  _flag8kHz = 0;
-    _flagG729mode = 0;
-    return;
+ACMG729_1::ACMG729_1(WebRtc_Word16 codecID)
+    : _encoderInstPtr(NULL),
+      _decoderInstPtr(NULL),
+      _myRate(32000),  // Default rate.
+      _flag8kHz(0),
+      _flagG729mode(0) {
+  // TODO(tlegrand): We should add codecID as a input variable to the
+  // constructor of ACMGenericCodec.
+  _codecID = codecID;
+  return;
 }
 
 ACMG729_1::~ACMG729_1()
diff --git a/src/modules/audio_coding/main/source/acm_speex.cc b/src/modules/audio_coding/main/source/acm_speex.cc
index 1f75e94ee..acb2a3b27 100644
--- a/src/modules/audio_coding/main/source/acm_speex.cc
+++ b/src/modules/audio_coding/main/source/acm_speex.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
@@ -54,7 +54,12 @@ namespace webrtc {
 #ifndef WEBRTC_CODEC_SPEEX
 ACMSPEEX::ACMSPEEX(WebRtc_Word16 /* codecID*/)
     : _encoderInstPtr(NULL),
-      _decoderInstPtr(NULL) {
+      _decoderInstPtr(NULL),
+      _complMode(0),
+      _vbrEnabled(false),
+      _encodingRate(-1),
+      _samplingFrequency(-1),
+      _samplesIn20MsAudio(-1) {
   return;
 }
 
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 77ed9ecf2..2abe46b7e 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
@@ -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
@@ -1191,9 +1191,8 @@ match");
     } else {
       // Copy payload data for future use.
       size_t length = static_cast<size_t>(
-          audioFrame._payloadDataLengthInSamples * audio_channels *
-          sizeof(WebRtc_UWord16));
-      memcpy(audio, audioFrame._payloadData, length);
+          audioFrame._payloadDataLengthInSamples * audio_channels);
+      memcpy(audio, audioFrame._payloadData, length * sizeof(WebRtc_UWord16));
     }
 
     WebRtc_UWord32 currentTimestamp;