diff --git a/src/modules/audio_coding/main/source/acm_celt.cc b/src/modules/audio_coding/main/source/acm_celt.cc
new file mode 100644
index 000000000..1e467fe89
--- /dev/null
+++ b/src/modules/audio_coding/main/source/acm_celt.cc
@@ -0,0 +1,317 @@
+/*
+ *  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
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "acm_common_defs.h"
+#include "acm_neteq.h"
+#include "acm_celt.h"
+#include "trace.h"
+#include "webrtc_neteq.h"
+#include "webrtc_neteq_help_macros.h"
+// TODO(tlegrand): Add full paths.
+
+#ifdef WEBRTC_CODEC_CELT
+// NOTE! Celt is not included in the open-source package. Modify this file or
+// your codec API to match the function call and name of used Celt API file.
+#include "celt_interface.h"
+#endif
+
+namespace webrtc {
+
+#ifndef WEBRTC_CODEC_CELT
+
+ACMCELT::ACMCELT(int16_t /* codecID */)
+    : enc_inst_ptr_(NULL),
+      dec_inst_ptr_(NULL),
+      sampling_freq_(0),
+      bitrate_(0),
+      channels_(1),
+      dec_channels_(1) {
+  return;
+}
+
+ACMCELT::~ACMCELT() {
+  return;
+}
+
+int16_t ACMCELT::InternalEncode(uint8_t* /* bitStream */,
+                                int16_t* /* bitStreamLenByte */) {
+  return -1;
+}
+
+int16_t ACMCELT::DecodeSafe(uint8_t* /* bitStream */,
+                            int16_t /* bitStreamLenByte */,
+                            int16_t* /* audio */,
+                            int16_t* /* audioSamples */,
+                            char* /* speechType */) {
+  return -1;
+}
+
+int16_t ACMCELT::InternalInitEncoder(WebRtcACMCodecParams* /* codecParams */) {
+  return -1;
+}
+
+int16_t ACMCELT::InternalInitDecoder(WebRtcACMCodecParams* /* codecParams */) {
+  return -1;
+}
+
+int32_t ACMCELT::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef  */,
+                          const CodecInst& /* codecInst */) {
+  return -1;
+}
+
+ACMGenericCodec* ACMCELT::CreateInstance(void) {
+  return NULL;
+}
+
+int16_t ACMCELT::InternalCreateEncoder() {
+  return -1;
+}
+
+void ACMCELT::DestructEncoderSafe() {
+  return;
+}
+
+int16_t ACMCELT::InternalCreateDecoder() {
+  return -1;
+}
+
+void ACMCELT::DestructDecoderSafe() {
+  return;
+}
+
+void ACMCELT::InternalDestructEncoderInst(void* /* ptrInst */) {
+  return;
+}
+
+int16_t ACMCELT::UnregisterFromNetEqSafe(ACMNetEQ* /* netEq */,
+                                         int16_t /* payloadType */) {
+  return -1;
+}
+
+bool ACMCELT::IsTrueStereoCodec() {
+  return true;
+}
+
+int16_t ACMCELT::SetBitRateSafe(const int32_t /*rate*/) {
+  return -1;
+}
+
+#else  //===================== Actual Implementation =======================
+
+ACMCELT::ACMCELT(int16_t codecID)
+    : enc_inst_ptr_(NULL),
+      dec_inst_ptr_(NULL),
+      sampling_freq_(32000),  // Default sampling frequency.
+      bitrate_(48000),  // Default rate.
+      channels_(1),  // Default send mono.
+      dec_channels_(1) {  // Default receive mono.
+  // TODO(tlegrand): remove later when ACMGenericCodec has a new constructor.
+  _codecID = codecID;
+
+  return;
+}
+
+ACMCELT::~ACMCELT() {
+  if (enc_inst_ptr_ != NULL) {
+    WebRtcCelt_FreeEnc(enc_inst_ptr_);
+    enc_inst_ptr_ = NULL;
+  }
+  if (dec_inst_ptr_ != NULL) {
+    WebRtcCelt_FreeDec(dec_inst_ptr_);
+    dec_inst_ptr_ = NULL;
+  }
+  return;
+}
+
+int16_t ACMCELT::InternalEncode(uint8_t* bitStream, int16_t* bitStreamLenByte) {
+  *bitStreamLenByte = 0;
+
+  // Call Encoder.
+  *bitStreamLenByte = WebRtcCelt_Encode(enc_inst_ptr_,
+                                        &_inAudio[_inAudioIxRead],
+                                        bitStream);
+
+  // Increment the read index this tell the caller that how far
+  // we have gone forward in reading the audio buffer.
+  _inAudioIxRead += _frameLenSmpl * channels_;
+
+  if (*bitStreamLenByte < 0) {
+    // Error reported from the encoder.
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+                 "InternalEncode: Encode error for Celt");
+    *bitStreamLenByte = 0;
+    return -1;
+  }
+
+  return *bitStreamLenByte;
+}
+
+int16_t ACMCELT::DecodeSafe(uint8_t* /* bitStream */,
+                            int16_t /* bitStreamLenByte */,
+                            int16_t* /* audio */,
+                            int16_t* /* audioSamples */,
+                            char* /* speechType */) {
+  return 0;
+}
+
+int16_t ACMCELT::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
+  // Set bitrate and check that it is withing the valid range.
+  int16_t status = SetBitRateSafe((codecParams->codecInstant).rate);
+  if (status < 0) {
+    return -1;
+  }
+
+  // If number of channels changed we need to re-create memory.
+  if (codecParams->codecInstant.channels != channels_) {
+    WebRtcCelt_FreeEnc(enc_inst_ptr_);
+    enc_inst_ptr_ = NULL;
+    // Store new number of channels.
+    channels_ = codecParams->codecInstant.channels;
+    if (WebRtcCelt_CreateEnc(&enc_inst_ptr_, channels_) < 0) {
+       return -1;
+    }
+  }
+
+  // Initiate encoder.
+  if (WebRtcCelt_EncoderInit(enc_inst_ptr_, channels_, bitrate_) >= 0) {
+    return 0;
+  } else {
+    return -1;
+  }
+}
+
+int16_t ACMCELT::InternalInitDecoder(WebRtcACMCodecParams* codecParams) {
+  // If number of channels changed we need to re-create memory.
+  if (codecParams->codecInstant.channels != dec_channels_) {
+    WebRtcCelt_FreeDec(dec_inst_ptr_);
+    dec_inst_ptr_ = NULL;
+    // Store new number of channels.
+    dec_channels_ = codecParams->codecInstant.channels;
+    if (WebRtcCelt_CreateDec(&dec_inst_ptr_, dec_channels_) < 0) {
+       return -1;
+    }
+  }
+
+  // Initiate decoder.
+  if (WebRtcCelt_DecoderInit(dec_inst_ptr_) < 0) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+                 "InternalInitDecoder: init decoder failed for Celt.");
+    return -1;
+  }
+  return 0;
+}
+
+int32_t ACMCELT::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
+                          const CodecInst& codecInst) {
+  if (!_decoderInitialized) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+                 "CodecDef: Decoder uninitialized for Celt");
+    return -1;
+  }
+
+  // Fill up the structure by calling
+  // "SET_CODEC_PAR" and "SET_CELT_FUNCTIONS" or "SET_CELTSLAVE_FUNCTIONS".
+  // Then call NetEQ to add the codec to it's
+  // database.
+  SET_CODEC_PAR((codecDef), kDecoderCELT_32, codecInst.pltype, dec_inst_ptr_,
+                32000);
+  // If this is the master of NetEQ, regular decoder will be added, otherwise
+  // the slave decoder will be used.
+  if (_isMaster) {
+    SET_CELT_FUNCTIONS((codecDef));
+  } else {
+    SET_CELTSLAVE_FUNCTIONS((codecDef));
+  }
+  return 0;
+}
+
+ACMGenericCodec* ACMCELT::CreateInstance(void) {
+  return NULL;
+}
+
+int16_t ACMCELT::InternalCreateEncoder() {
+  if (WebRtcCelt_CreateEnc(&enc_inst_ptr_, _noChannels) < 0) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+                 "InternalCreateEncoder: create encoder failed for Celt");
+    return -1;
+  }
+  channels_ = _noChannels;
+  return 0;
+}
+
+void ACMCELT::DestructEncoderSafe() {
+  _encoderExist = false;
+  _encoderInitialized = false;
+  if (enc_inst_ptr_ != NULL) {
+    WebRtcCelt_FreeEnc(enc_inst_ptr_);
+    enc_inst_ptr_ = NULL;
+  }
+}
+
+int16_t ACMCELT::InternalCreateDecoder() {
+  if (WebRtcCelt_CreateDec(&dec_inst_ptr_, dec_channels_) < 0) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+                 "InternalCreateDecoder: create decoder failed for Celt");
+    return -1;
+  }
+
+  return 0;
+}
+
+void ACMCELT::DestructDecoderSafe() {
+  _decoderExist = false;
+  _decoderInitialized = false;
+  if (dec_inst_ptr_ != NULL) {
+    WebRtcCelt_FreeDec(dec_inst_ptr_);
+    dec_inst_ptr_ = NULL;
+  }
+}
+
+void ACMCELT::InternalDestructEncoderInst(void* ptrInst) {
+  if (ptrInst != NULL) {
+    WebRtcCelt_FreeEnc(static_cast<CELT_encinst_t*>(ptrInst));
+  }
+  return;
+}
+
+int16_t ACMCELT::UnregisterFromNetEqSafe(ACMNetEQ* netEq, int16_t payloadType) {
+  if (payloadType != _decoderParams.codecInstant.pltype) {
+    WEBRTC_TRACE(
+        webrtc::kTraceError,
+        webrtc::kTraceAudioCoding,
+        _uniqueID,
+        "Cannot unregister codec: given payload-type does not match the stored "
+        "payload type",
+        _decoderParams.codecInstant.plname, payloadType,
+        _decoderParams.codecInstant.pltype);
+    return -1;
+  }
+  return netEq->RemoveCodec(kDecoderCELT_32);
+}
+
+bool ACMCELT::IsTrueStereoCodec() {
+  return true;
+}
+
+int16_t ACMCELT::SetBitRateSafe(const int32_t rate) {
+  // Check that rate is in the valid range.
+  if ((rate >= 48000) && (rate <= 96000)) {
+    bitrate_ = rate;
+    return 0;
+  } else {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
+                 "SetBitRateSafe: Invalid rate Celt");
+    return -1;
+  }
+}
+
+#endif
+
+}  // namespace webrtc
diff --git a/src/modules/audio_coding/main/source/acm_celt.h b/src/modules/audio_coding/main/source/acm_celt.h
new file mode 100644
index 000000000..2797a9e43
--- /dev/null
+++ b/src/modules/audio_coding/main/source/acm_celt.h
@@ -0,0 +1,72 @@
+/*
+ *  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
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
+
+#include "acm_generic_codec.h"
+
+// forward declaration
+struct CELT_encinst_t_;
+struct CELT_decinst_t_;
+
+namespace webrtc {
+
+class ACMCELT : public ACMGenericCodec {
+ public:
+  ACMCELT(int16_t codecID);
+  ~ACMCELT();
+
+  ACMGenericCodec* CreateInstance(void);
+
+  int16_t InternalEncode(uint8_t* bitstream, int16_t* bitStreamLenByte);
+
+  int16_t InternalInitEncoder(WebRtcACMCodecParams *codecParams);
+
+  int16_t InternalInitDecoder(WebRtcACMCodecParams *codecParams);
+
+ protected:
+
+  WebRtc_Word16 DecodeSafe(
+      WebRtc_UWord8* bitStream,
+      WebRtc_Word16  bitStreamLenByte,
+      WebRtc_Word16* audio,
+      WebRtc_Word16* audioSamples,
+      WebRtc_Word8*  speechType);
+
+  int32_t CodecDef(WebRtcNetEQ_CodecDef& codecDef, const CodecInst& codecInst);
+
+  void DestructEncoderSafe();
+
+  void DestructDecoderSafe();
+
+  int16_t InternalCreateEncoder();
+
+  int16_t InternalCreateDecoder();
+
+  void InternalDestructEncoderInst(void* ptrInst);
+
+  int16_t UnregisterFromNetEqSafe(ACMNetEQ* netEq, int16_t payloadType);
+
+  bool IsTrueStereoCodec();
+
+  int16_t SetBitRateSafe(const int32_t rate);
+
+  CELT_encinst_t_* enc_inst_ptr_;
+  CELT_decinst_t_* dec_inst_ptr_;
+  uint16_t sampling_freq_;
+  uint16_t bitrate_;
+  uint16_t channels_;
+  uint16_t dec_channels_;
+};
+
+} // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
diff --git a/src/modules/audio_coding/main/source/acm_codec_database.cc b/src/modules/audio_coding/main/source/acm_codec_database.cc
index 9da92b18c..d75414df9 100644
--- a/src/modules/audio_coding/main/source/acm_codec_database.cc
+++ b/src/modules/audio_coding/main/source/acm_codec_database.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
@@ -58,6 +58,10 @@
     #include "acm_amrwb.h"
     #include "amrwb_interface.h"
 #endif
+#ifdef WEBRTC_CODEC_CELT
+    #include "acm_celt.h"
+    #include "celt_interface.h"
+#endif
 #ifdef WEBRTC_CODEC_G722
     #include "acm_g722.h"
     #include "g722_interface.h"
@@ -111,8 +115,9 @@ const int kDynamicPayloadtypes[ACMCodecDB::kMaxNumCodecs] = {
 // default channel support, and default rate.
 #if (defined(WEBRTC_CODEC_PCM16) || \
      defined(WEBRTC_CODEC_AMR) || defined(WEBRTC_CODEC_AMRWB) || \
-     defined(WEBRTC_CODEC_G729_1) || defined(WEBRTC_CODEC_SPEEX) || \
-     defined(WEBRTC_CODEC_G722_1) || defined(WEBRTC_CODEC_G722_1C))
+     defined(WEBRTC_CODEC_CELT) || defined(WEBRTC_CODEC_G729_1) || \
+     defined(WEBRTC_CODEC_SPEEX) || defined(WEBRTC_CODEC_G722_1) || \
+     defined(WEBRTC_CODEC_G722_1C))
 static int count_database = 0;
 #endif
 
@@ -140,6 +145,9 @@ const CodecInst ACMCodecDB::database_[] = {
 #ifdef WEBRTC_CODEC_AMRWB
   {kDynamicPayloadtypes[count_database++], "AMR-WB", 16000, 320, 1, 20000},
 #endif
+#ifdef WEBRTC_CODEC_CELT
+  {kDynamicPayloadtypes[count_database++], "CELT", 32000, 320, 1, 48000},
+#endif
 #ifdef WEBRTC_CODEC_G722
   {9, "G722", 16000, 320, 1, 64000},
 #endif
@@ -208,6 +216,9 @@ const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
 #ifdef WEBRTC_CODEC_AMRWB
   {3, {320, 640, 960}, 0, 1},
 #endif
+#ifdef WEBRTC_CODEC_CELT
+  {1, {320}, 0, 2},
+#endif
 #ifdef WEBRTC_CODEC_G722
   {6, {160, 320, 480, 640, 800, 960}, 0, 2},
 #endif
@@ -273,6 +284,9 @@ const WebRtcNetEQDecoder ACMCodecDB::neteq_decoders_[] = {
 #ifdef WEBRTC_CODEC_AMRWB
   kDecoderAMRWB,
 #endif
+#ifdef WEBRTC_CODEC_CELT
+  kDecoderCELT_32,
+#endif
 #ifdef WEBRTC_CODEC_G722
   kDecoderG722,
 #endif
@@ -698,6 +712,10 @@ ACMGenericCodec* ACMCodecDB::CreateCodecInstance(const CodecInst* codec_inst) {
   } else if (!STR_CASE_CMP(codec_inst->plname, "AMR-WB")) {
 #ifdef WEBRTC_CODEC_AMRWB
     return new ACMAMRwb(kGSMAMRWB);
+#endif
+  } else if (!STR_CASE_CMP(codec_inst->plname, "CELT")) {
+#ifdef WEBRTC_CODEC_CELT
+    return new ACMCELT(kCELT32);
 #endif
   } else if (!STR_CASE_CMP(codec_inst->plname, "G722")) {
 #ifdef WEBRTC_CODEC_G722
diff --git a/src/modules/audio_coding/main/source/acm_codec_database.h b/src/modules/audio_coding/main/source/acm_codec_database.h
index 229d4a8a5..1752bbc40 100644
--- a/src/modules/audio_coding/main/source/acm_codec_database.h
+++ b/src/modules/audio_coding/main/source/acm_codec_database.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
@@ -51,6 +51,9 @@ class ACMCodecDB {
 #ifdef WEBRTC_CODEC_AMRWB
     , kGSMAMRWB
 #endif
+#ifdef WEBRTC_CODEC_CELT
+    , kCELT32
+#endif
 #ifdef WEBRTC_CODEC_G722
     , kG722
 #endif
@@ -112,6 +115,9 @@ class ACMCodecDB {
 #ifndef WEBRTC_CODEC_AMRWB
   enum {kGSMAMRWB = -1};
 #endif
+#ifndef WEBRTC_CODEC_CELT
+  enum {kCELT32 = -1};
+#endif
 #ifndef WEBRTC_CODEC_G722
   enum {kG722 = -1};
 #endif
diff --git a/src/modules/audio_coding/main/source/acm_generic_codec.cc b/src/modules/audio_coding/main/source/acm_generic_codec.cc
index 44aad87ef..0e212fb29 100644
--- a/src/modules/audio_coding/main/source/acm_generic_codec.cc
+++ b/src/modules/audio_coding/main/source/acm_generic_codec.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
@@ -664,6 +664,7 @@ ACMGenericCodec::InitEncoderSafe(
         }
     }
     _frameLenSmpl = (codecParams->codecInstant).pacsize;
+    _noChannels = codecParams->codecInstant.channels;
     status = InternalInitEncoder(codecParams);
     if(status < 0)
     {
@@ -700,8 +701,6 @@ ACMGenericCodec::InitEncoderSafe(
     status = SetVADSafe(codecParams->enableDTX, codecParams->enableVAD,
         codecParams->vadMode);
 
-    _noChannels = codecParams->codecInstant.channels;
-
     return status;
 }
 
diff --git a/src/modules/audio_coding/main/source/acm_generic_codec.h b/src/modules/audio_coding/main/source/acm_generic_codec.h
index 7f50669ea..7f8ef8c47 100644
--- a/src/modules/audio_coding/main/source/acm_generic_codec.h
+++ b/src/modules/audio_coding/main/source/acm_generic_codec.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
@@ -811,6 +811,18 @@ public:
         WebRtc_UWord8*      payload,
         WebRtc_Word16*      payloadLenBytes);
 
+    ///////////////////////////////////////////////////////////////////////////
+    // IsTrueStereoCodec()
+    // Call to see if current encoder is a true stereo codec. This function
+    // should be overwritten for codecs which are true stereo codecs
+    // Return value:
+    //   -true  if stereo codec
+    //   -false if not stereo codec.
+    //
+    virtual bool IsTrueStereoCodec() {
+      return false;
+    }
+
 protected:
     ///////////////////////////////////////////////////////////////////////////
     // All the functions with FunctionNameSafe(...) contain the actual
@@ -1262,7 +1274,6 @@ protected:
     virtual void SaveDecoderParamSafe(
         const WebRtcACMCodecParams* codecParams);
 
-
     // &_inAudio[_inAudioIxWrite] always point to where new audio can be
     // written to
     WebRtc_Word16         _inAudioIxWrite;
diff --git a/src/modules/audio_coding/main/source/audio_coding_module.gypi b/src/modules/audio_coding/main/source/audio_coding_module.gypi
index 39434e519..80114e0ae 100644
--- a/src/modules/audio_coding/main/source/audio_coding_module.gypi
+++ b/src/modules/audio_coding/main/source/audio_coding_module.gypi
@@ -1,4 +1,4 @@
-# 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
@@ -42,6 +42,8 @@
         'acm_amr.h',
         'acm_amrwb.cc',
         'acm_amrwb.h',
+        'acm_celt.cc',
+        'acm_celt.h',
         'acm_cng.cc',
         'acm_cng.h',
         'acm_codec_database.cc',
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 c48c4c408..6691fa3ce 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
@@ -1642,6 +1642,12 @@ AudioCodingModuleImpl::RegisterRecCodecMSSafe(
     else if(jitterBuffer == ACMNetEQ::slaveJB)
     {
         codecArray = &_slaveCodecs[0];
+        if (_codecs[codecId]->IsTrueStereoCodec()) {
+          // True stereo codecs need to use the same codec memory
+          // for both master and slave.
+          _slaveCodecs[mirrorId] = _codecs[mirrorId];
+          _mirrorCodecIdx[mirrorId] = mirrorId;
+        }
     }
     else
     {
@@ -1827,6 +1833,8 @@ AudioCodingModuleImpl::IncomingPacket(
                         // current payload type.
                         if (_stereoReceive[i]) {
                           _expected_channels = 2;
+                        } else {
+                          _expected_channels = 1;
                         }
 
                         // Reset previous received channel
diff --git a/src/modules/audio_coding/main/test/TestAllCodecs.cc b/src/modules/audio_coding/main/test/TestAllCodecs.cc
index 2ad3e1ec0..c64e269e3 100644
--- a/src/modules/audio_coding/main/test/TestAllCodecs.cc
+++ b/src/modules/audio_coding/main/test/TestAllCodecs.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
@@ -583,7 +583,7 @@ void TestAllCodecs::Perform()
     } else {
         printf(".");
     }
-    _testCntr++; 
+    _testCntr++;
     OpenOutFile(_testCntr);
     char codecPCMA[] = "PCMA";
     RegisterSendCodec('A', codecPCMA, 8000, 64000, 80, 0);
@@ -597,13 +597,12 @@ void TestAllCodecs::Perform()
     RegisterSendCodec('A', codecPCMA, 8000, 64000, 400, 0);
     Run(_channelA2B);
     RegisterSendCodec('A', codecPCMA, 8000, 64000, 480, 0);
+    Run(_channelA2B);
     if(_testMode != 0) {
         printf("=======================================================================\n");
     } else {
         printf(".");
     }
-    _testCntr++;
-    Run(_channelA2B);
     char codecPCMU[] = "PCMU";
     RegisterSendCodec('A', codecPCMU, 8000, 64000, 80, 0);
     Run(_channelA2B);
@@ -650,7 +649,19 @@ void TestAllCodecs::Perform()
     Run(_channelA2B);
     _outFileB.Close();
 #endif
-
+#ifdef WEBRTC_CODEC_CELT
+    if(_testMode != 0) {
+        printf("=======================================================================\n");
+    } else {
+        printf(".");
+    }
+    _testCntr++;
+    OpenOutFile(_testCntr);
+    char codecCELT_32[] = "CELT";
+    RegisterSendCodec('A', codecCELT_32, 32000, 48000, 320, 0);
+    Run(_channelA2B);
+    _outFileB.Close();
+#endif
     if(_testMode != 0) {
         printf("=======================================================================\n");
     } else {
diff --git a/src/modules/audio_coding/main/test/TestStereo.cc b/src/modules/audio_coding/main/test/TestStereo.cc
index ef633de66..0b267ff39 100644
--- a/src/modules/audio_coding/main/test/TestStereo.cc
+++ b/src/modules/audio_coding/main/test/TestStereo.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
@@ -104,6 +104,11 @@ TestPackStereo::SendData(
                 payloadDataMaster[j] = (payloadData[i] & 0xF0) + (payloadData[i+1] >> 4);
                 payloadDataSlave[j] = ((payloadData[i] & 0x0F) << 4) + (payloadData[i+1] & 0x0F);
             }
+        } else if (_codecType == 4) {
+          // True stereo, call both master and slave with whole stream.
+          memcpy(payloadDataMaster, payloadData, payloadSize);
+          memcpy(payloadDataSlave, payloadData, payloadSize);
+          payloadDataSize = payloadSize*2;
         }
     }
     else
@@ -115,7 +120,7 @@ TestPackStereo::SendData(
         payloadDataSize = payloadSize*2;
     }
 
-    if (_codecType != 4) {
+    if (_codecType != 5) {
       // Call ACM with two packets, one for each channel
       rtpInfo.type.Audio.channel = 1;
       status = _receiverACM->IncomingPacket((WebRtc_Word8*)payloadDataMaster, payloadDataSize/2, rtpInfo);
@@ -235,7 +240,8 @@ void TestStereo::Perform()
         if(!strcmp(myCodecParam.plname, "L16") || 
             !strcmp(myCodecParam.plname, "PCMA")|| 
             !strcmp(myCodecParam.plname, "PCMU")|| 
-            !strcmp(myCodecParam.plname, "G722"))
+            !strcmp(myCodecParam.plname, "G722")||
+            !strcmp(myCodecParam.plname, "CELT"))
         {
             myCodecParam.channels=2;
             _acmB->RegisterReceiveCodec(myCodecParam);
@@ -246,7 +252,7 @@ void TestStereo::Perform()
     _channelA2B = new TestPackStereo;    
     _acmA->RegisterTransportCallback(_channelA2B);
     _channelA2B->RegisterReceiverACM(_acmB);
-    
+
     //
     // Test Stereo-To-Stereo for all codecs.
     //
@@ -410,10 +416,30 @@ void TestStereo::Perform()
     RegisterSendCodec('A', codecPCMU, 8000, 64000, 80, codec_channels);
     Run(_channelA2B, audio_channels, codec_channels);
     _acmA->SetVAD(false, false, VADNormal);
-    Run(_channelA2B, audio_channels, codec_channels);
     _outFileB.Close();
 #endif
-
+#ifdef WEBRTC_CODEC_CELT
+    if(_testMode != 0) {
+        printf("=======================================================================\n");
+        printf("Test number: %d\n",_testCntr + 1);
+        printf("Test type: Stereo-to-stereo\n");
+    } else {
+        printf(".");
+    }
+    _channelA2B->SetCodecType(4);
+    audio_channels = 2;
+    codec_channels = 2;
+    _testCntr++;
+    OpenOutFile(_testCntr);
+    char codecCELT[] = "CELT";
+    RegisterSendCodec('A', codecCELT, 32000, 48000, 320, codec_channels);
+    Run(_channelA2B, audio_channels, codec_channels);
+    _acmA->SetVAD(true, true, VADNormal);
+    RegisterSendCodec('A', codecCELT, 32000, 48000, 320, codec_channels);
+    Run(_channelA2B, audio_channels, codec_channels);
+    _acmA->SetVAD(false, false, VADNormal);
+    _outFileB.Close();
+#endif
   //
   // Test Mono-To-Stereo for all codecs.
   //
@@ -481,13 +507,26 @@ void TestStereo::Perform()
   Run(_channelA2B, audio_channels, codec_channels);
   _outFileB.Close();
 #endif
+#ifdef WEBRTC_CODEC_CELT
+  if(_testMode != 0) {
+    printf("===============================================================\n");
+    printf("Test number: %d\n",_testCntr + 1);
+    printf("Test type: Mono-to-stereo\n");
+  }
+  _testCntr++;
+  _channelA2B->SetCodecType(4);
+  OpenOutFile(_testCntr);
+  RegisterSendCodec('A', codecCELT, 32000, 48000, 320, codec_channels);
+  Run(_channelA2B, audio_channels, codec_channels);
+  _outFileB.Close();
+#endif
 
   //
   // Test Stereo-To-Mono for all codecs.
   //
   audio_channels = 2;
   codec_channels = 1;
-  _channelA2B->SetCodecType(4);
+  _channelA2B->SetCodecType(5);
 
   // Register receivers as mono.
   for(WebRtc_UWord8 n = 0; n < numEncoders; n++) {
@@ -495,7 +534,8 @@ void TestStereo::Perform()
     if(!strcmp(myCodecParam.plname, "L16") ||
         !strcmp(myCodecParam.plname, "PCMA")||
         !strcmp(myCodecParam.plname, "PCMU")||
-        !strcmp(myCodecParam.plname, "G722")) {
+        !strcmp(myCodecParam.plname, "G722")||
+        !strcmp(myCodecParam.plname, "CELT")) {
       myCodecParam.channels = 1;
       _acmB->RegisterReceiveCodec(myCodecParam);
     }
@@ -537,7 +577,7 @@ void TestStereo::Perform()
   if(_testMode != 0) {
      printf("===============================================================\n");
      printf("Test number: %d\n",_testCntr + 1);
-     printf("Test type: Mono-to-stereo\n");
+     printf("Test type: Stereo-to-mono\n");
    }
    _testCntr++;
    OpenOutFile(_testCntr);
@@ -549,7 +589,7 @@ void TestStereo::Perform()
   if(_testMode != 0) {
     printf("===============================================================\n");
     printf("Test number: %d\n",_testCntr + 1);
-    printf("Test type: Mono-to-stereo\n");
+    printf("Test type: Stereo-to-mono\n");
   }
   _testCntr++;
   OpenOutFile(_testCntr);
@@ -559,10 +599,22 @@ void TestStereo::Perform()
   Run(_channelA2B, audio_channels, codec_channels);
   _outFileB.Close();
 #endif
+#ifdef WEBRTC_CODEC_CELT
+  if(_testMode != 0) {
+    printf("===============================================================\n");
+    printf("Test number: %d\n",_testCntr + 1);
+    printf("Test type: Stereo-to-mono\n");
+  }
+  _testCntr++;
+  OpenOutFile(_testCntr);
+  RegisterSendCodec('A', codecCELT, 32000, 48000, 320, codec_channels);
+  Run(_channelA2B, audio_channels, codec_channels);
+  _outFileB.Close();
+#endif
 
     // Print out which codecs were tested, and which were not, in the run.
     if(_testMode != 0) {
-        printf("/nThe following codecs was INCLUDED in the test:\n");
+        printf("\nThe following codecs was INCLUDED in the test:\n");
 #ifdef WEBRTC_CODEC_G722
         printf("   G.722\n");
 #endif
@@ -716,6 +768,9 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels)
     if (_counter == 1000) {
         _counter = 0;
     }
+    if (_in_file_mono.EndOfFile()) {
+        _in_file_mono.Rewind();
+    }
     if (_in_file_stereo.EndOfFile()) {
         _in_file_stereo.Rewind();
     }
diff --git a/src/modules/audio_coding/neteq/codec_db.c b/src/modules/audio_coding/neteq/codec_db.c
index c15306d3b..cb63aea6c 100644
--- a/src/modules/audio_coding/neteq/codec_db.c
+++ b/src/modules/audio_coding/neteq/codec_db.c
@@ -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
@@ -161,6 +161,9 @@ int WebRtcNetEQ_DbAdd(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec,
         case kDecoderSPEEX_8 :
         case kDecoderSPEEX_16 :
 #endif
+#ifdef NETEQ_CELT_CODEC
+        case kDecoderCELT_32 :
+#endif
 #ifdef NETEQ_GSMFR_CODEC
         case kDecoderGSMFR :
 #endif
@@ -468,6 +471,9 @@ int WebRtcNetEQ_DbGetSplitInfo(SplitInfo_t *inst, enum WebRtcNetEQDecoder codecI
         case kDecoderSPEEX_8:
         case kDecoderSPEEX_16:
 #endif
+#ifdef NETEQ_CELT_CODEC
+        case kDecoderCELT_32 :
+#endif
 #ifdef NETEQ_G729_1_CODEC
         case kDecoderG729_1:
 #endif
diff --git a/src/modules/audio_coding/neteq/interface/webrtc_neteq.h b/src/modules/audio_coding/neteq/interface/webrtc_neteq.h
index 744bd624c..4fef5531c 100644
--- a/src/modules/audio_coding/neteq/interface/webrtc_neteq.h
+++ b/src/modules/audio_coding/neteq/interface/webrtc_neteq.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
@@ -58,6 +58,7 @@ enum WebRtcNetEQDecoder
     kDecoderG722_1C_48,
     kDecoderSPEEX_8,
     kDecoderSPEEX_16,
+    kDecoderCELT_32,
     kDecoderGSMFR,
     kDecoderAMR,
     kDecoderAMRWB,
diff --git a/src/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h b/src/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h
index c63b3464d..214bd10d4 100644
--- a/src/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h
+++ b/src/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.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
@@ -328,6 +328,28 @@
                     inst.funcUpdBWEst=NULL; \
                     inst.funcGetErrorCode=NULL;
 
+#define SET_CELT_FUNCTIONS(inst) \
+                    inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcCelt_Decode; \
+                    inst.funcDecodeRCU=NULL; \
+                    inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcCelt_DecodePlc; \
+                    inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcCelt_DecoderInit; \
+                    inst.funcAddLatePkt=NULL; \
+                    inst.funcGetMDinfo=NULL; \
+                    inst.funcGetPitch=NULL; \
+                    inst.funcUpdBWEst=NULL; \
+                    inst.funcGetErrorCode=NULL;
+
+#define SET_CELTSLAVE_FUNCTIONS(inst) \
+                    inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcCelt_DecodeSlave; \
+                    inst.funcDecodeRCU=NULL; \
+                    inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcCelt_DecodePlc; \
+                    inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcCelt_DecoderInit; \
+                    inst.funcAddLatePkt=NULL; \
+                    inst.funcGetMDinfo=NULL; \
+                    inst.funcGetPitch=NULL; \
+                    inst.funcUpdBWEst=NULL; \
+                    inst.funcGetErrorCode=NULL;
+
 #define SET_RED_FUNCTIONS(inst) \
                     inst.funcDecode=NULL; \
                     inst.funcDecodeRCU=NULL; \
diff --git a/src/modules/audio_coding/neteq/neteq_defines.h b/src/modules/audio_coding/neteq/neteq_defines.h
index e428b8b4e..318e6bb7f 100644
--- a/src/modules/audio_coding/neteq/neteq_defines.h
+++ b/src/modules/audio_coding/neteq/neteq_defines.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
@@ -79,6 +79,8 @@
  *
  * NETEQ_SPEEX_CODEC              Enable Speex (at 8 and 16 kHz sample rate)
  *
+ * NETEQ_CELT_CODEC               Enable Celt (at 32 kHz sample rate)
+ *
  * NETEQ_GSMFR_CODEC              Enable GSM-FR
  *
  * NETEQ_AMR_CODEC                Enable AMR (narrowband)
@@ -259,6 +261,7 @@
     #define NETEQ_ISAC_SWB_CODEC
     #define NETEQ_32KHZ_WIDEBAND
     #define NETEQ_G722_1C_CODEC
+    #define NETEQ_CELT_CODEC
 #endif
 
 #if (defined(NETEQ_VOICEENGINE_CODECS))
@@ -290,6 +293,7 @@
     #define NETEQ_ISAC_SWB_CODEC
     #define NETEQ_32KHZ_WIDEBAND
     #define NETEQ_G722_1C_CODEC
+    #define NETEQ_CELT_CODEC
 
 #endif 
 
@@ -323,6 +327,7 @@
     #define NETEQ_ISAC_SWB_CODEC
     #define NETEQ_32KHZ_WIDEBAND
     #define NETEQ_G722_1C_CODEC
+    #define NETEQ_CELT_CODEC
 
     /* Super wideband 48kHz codecs */
     #define NETEQ_48KHZ_WIDEBAND
diff --git a/src/modules/audio_coding/neteq/packet_buffer.c b/src/modules/audio_coding/neteq/packet_buffer.c
index 4b69704ed..8b9073c09 100644
--- a/src/modules/audio_coding/neteq/packet_buffer.c
+++ b/src/modules/audio_coding/neteq/packet_buffer.c
@@ -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
@@ -678,6 +678,11 @@ int WebRtcNetEQ_GetDefaultCodecSettings(const enum WebRtcNetEQDecoder *codecID,
             codecBytes = 1250; /* 210ms @ 50kbps */
             codecBuffers = 10;
         }
+        else if (codecID[i] == kDecoderCELT_32)
+        {
+            codecBytes = 1250; /* 210ms @ 50kbps */
+            codecBuffers = 10;
+        }
         else if (codecID[i] == kDecoderGSMFR)
         {
             codecBytes = 340; /* 200ms */
diff --git a/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.cc b/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.cc
index 8956a20f0..64bf508f3 100644
--- a/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.cc
+++ b/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.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
@@ -580,6 +580,31 @@ int decoder_SPEEX::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
 }
 #endif
 
+#ifdef CODEC_CELT_32
+#include "celt_interface.h"
+decoder_CELT::decoder_CELT(WebRtc_UWord8 pt, WebRtc_UWord16 fs)
+:
+NETEQTEST_Decoder(kDecoderCELT_32, fs, "CELT", pt)
+{
+   if (WebRtcCelt_CreateDec((CELT_decinst_t **) &_decoder, 1))
+        exit(EXIT_FAILURE);
+}
+
+decoder_CELT::~decoder_CELT()
+{
+    WebRtcCelt_FreeDec((CELT_decinst_t *) _decoder);
+}
+
+int decoder_CELT::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
+{
+    WebRtcNetEQ_CodecDef codecInst;
+
+    SET_CELT_FUNCTIONS(codecInst);
+
+    return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
+}
+#endif
+
 #ifdef CODEC_RED
 int decoder_RED::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
 {
diff --git a/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h b/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h
index 61896f4a0..ff4904929 100644
--- a/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h
+++ b/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.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
@@ -265,6 +265,14 @@ public:
     int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
 };
 
+class decoder_CELT : public NETEQTEST_Decoder
+{
+public:
+    decoder_CELT(WebRtc_UWord8 pt = 0, WebRtc_UWord16 fs = 32000);
+    virtual ~decoder_CELT();
+    int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
+};
+
 class decoder_RED : public NETEQTEST_Decoder
 {
 public:
diff --git a/src/modules/audio_coding/neteq/test/NetEqRTPplay.cc b/src/modules/audio_coding/neteq/test/NetEqRTPplay.cc
index 812beee28..d3f7eb511 100644
--- a/src/modules/audio_coding/neteq/test/NetEqRTPplay.cc
+++ b/src/modules/audio_coding/neteq/test/NetEqRTPplay.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
@@ -1027,6 +1027,12 @@ void parsePtypeFile(FILE *ptypeFile, std::map<WebRtc_UWord8, decoderStruct>* dec
                 tempDecoder.fs = 16000;
             }
 #endif
+#ifdef CODEC_CELT_32
+            else if(strcmp(codec, "celt32") == 0) {
+                tempDecoder.codec = kDecoderCELT_32;
+                tempDecoder.fs = 32000;
+            }
+#endif
 #ifdef CODEC_SILK_NB
             else if(strcmp(codec, "silk8") == 0) {
                 tempDecoder.codec = NETEQ_CODEC_SILK_8;
@@ -1421,6 +1427,11 @@ void createAndInsertDecoders (NETEQTEST_NetEQClass *neteq, std::map<WebRtc_UWord
                 *dec = new decoder_SPEEX( pt, 16000 );
                 break;
 #endif
+#ifdef CODEC_CELT_32
+            case kDecoderCELT_32:
+                *dec = new decoder_CELT( pt, 32000 );
+                break;
+#endif
 #ifdef CODEC_RED
             case kDecoderRED:
                 *dec = new decoder_RED( pt );
diff --git a/src/modules/audio_coding/neteq/test/PayloadTypes.h b/src/modules/audio_coding/neteq/test/PayloadTypes.h
index b11a344ab..f6cc3da80 100644
--- a/src/modules/audio_coding/neteq/test/PayloadTypes.h
+++ b/src/modules/audio_coding/neteq/test/PayloadTypes.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
@@ -54,6 +54,7 @@
 #define NETEQ_CODEC_G729_1_PT			107
 #define NETEQ_CODEC_G729D_PT			123
 #define NETEQ_CODEC_MELPE_PT			124
+#define NETEQ_CODEC_CELT32_PT     114
 
 /* Extra dynamic codepoints */
 #define NETEQ_CODEC_AMRWB_PT			120
diff --git a/src/modules/audio_coding/neteq/test/RTPencode.cc b/src/modules/audio_coding/neteq/test/RTPencode.cc
index 8d445a840..9efee9224 100644
--- a/src/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/src/modules/audio_coding/neteq/test/RTPencode.cc
@@ -155,6 +155,9 @@ void stereoInterleave(unsigned char* data, int dataLen, int stride);
 #if ((defined CODEC_SPEEX_8)||(defined CODEC_SPEEX_16))
 	#include "SpeexInterface.h"
 #endif
+#ifdef CODEC_CELT_32
+#include "celt_interface.h"
+#endif
 
 
 /***********************************/
@@ -227,6 +230,9 @@ WebRtcVadInst *VAD_inst[2];
 #ifdef CODEC_SPEEX_16
 	SPEEX_encinst_t *SPEEX16enc_inst[2];
 #endif
+#ifdef CODEC_CELT_32
+  CELT_encinst_t *CELT32enc_inst[2];
+#endif
 #ifdef CODEC_G711
     void *G711state[2]={NULL, NULL};
 #endif
@@ -368,6 +374,9 @@ int main(int argc, char* argv[])
 #ifdef CODEC_SPEEX_16
 		printf("             : speex16      speex coder (16 kHz)\n");
 #endif
+#ifdef CODEC_CELT_32
+    printf("             : celt32       celt coder (32 kHz)\n");
+#endif
 #ifdef CODEC_RED
 #ifdef CODEC_G711
 		printf("             : red_pcm      Redundancy RTP packet with 2*G711A frames\n");
@@ -950,6 +959,11 @@ void NetEQTest_GetCodec_and_PT(char * name, enum WebRtcNetEQDecoder *codec, int
 		*codec=kDecoderSPEEX_16;
 		*PT=NETEQ_CODEC_SPEEX16_PT;
 	}
+  else if(!strcmp(name,"celt32")){
+    *fs=32000;
+    *codec=kDecoderCELT_32;
+    *PT=NETEQ_CODEC_CELT32_PT;
+  }
     else if(!strcmp(name,"red_pcm")){
 		*codec=kDecoderPCMa;
 		*PT=NETEQ_CODEC_PCMA_PT; /* this will be the PT for the sub-headers */
@@ -1122,6 +1136,26 @@ int NetEQTest_init_coders(enum WebRtcNetEQDecoder coder, int enc_frameSize, int
         }
         break;
 #endif
+#ifdef CODEC_CELT_32
+    case kDecoderCELT_32 :
+        if (sampfreq==32000) {
+            if (enc_frameSize==320) {
+                ok=WebRtcCelt_CreateEnc(&CELT32enc_inst[k], 1 /*mono*/);
+                if (ok!=0) {
+                    printf("Error: Couldn't allocate memory for Celt encoding instance\n");
+                    exit(0);
+                }
+            } else {
+                printf("\nError: Celt only supports 10 ms!!\n\n");
+                exit(0);
+            }
+            ok=WebRtcCelt_EncoderInit(CELT32enc_inst[k],  1 /*mono*/, 48000 /*bitrate*/);
+            if (ok!=0) exit(0);
+        } else {
+          printf("\nError - Celt32 called with sample frequency other than 32 kHz.\n\n");
+        }
+        break;
+#endif
 
 #ifdef CODEC_G722_1_16
     case kDecoderG722_1_16 :
@@ -1511,6 +1545,11 @@ int NetEQTest_free_coders(enum WebRtcNetEQDecoder coder, int numChannels) {
             WebRtcSpeex_FreeEnc(SPEEX16enc_inst[k]);
             break;
 #endif
+#ifdef CODEC_CELT_32
+        case kDecoderCELT_32 :
+            WebRtcCelt_FreeEnc(CELT32enc_inst[k]);
+            break;
+#endif
 
 #ifdef CODEC_G722_1_16
         case kDecoderG722_1_16 :
@@ -1814,6 +1853,20 @@ int NetEQTest_encode(int coder, WebRtc_Word16 *indata, int frameLen, unsigned ch
             cdlen=WebRtcSpeex_GetBitstream(SPEEX16enc_inst[k], (WebRtc_Word16*)encoded);
         }
 #endif
+#ifdef CODEC_CELT_32
+        else if (coder==kDecoderCELT_32) { /* Celt */
+            int encodedLen = 0;
+            cdlen = 0;
+            while (cdlen <= 0) {
+                cdlen = WebRtcCelt_Encode(CELT32enc_inst[k], &indata[encodedLen], encoded);
+                encodedLen += 10*32; /* 10 ms */
+            }
+            if( (encodedLen != frameLen) || cdlen < 0) {
+                printf("Error encoding Celt frame!\n");
+                exit(0);
+            }
+        }
+#endif
 
         indata += frameLen;
         encoded += cdlen;
diff --git a/src/modules/audio_coding/neteq/test/ptypes.txt b/src/modules/audio_coding/neteq/test/ptypes.txt
index ea47568f9..c3d4e2572 100644
--- a/src/modules/audio_coding/neteq/test/ptypes.txt
+++ b/src/modules/audio_coding/neteq/test/ptypes.txt
@@ -17,3 +17,4 @@ pcm16b_swb32khz 95
 //pcm16b_swb48khz 96
 //mpeg4aac 122
 g722 9
+celt32 114
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index 80da50506..b61157d67 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -225,6 +225,7 @@ ModuleRTPUtility::Payload* RTPReceiverAudio::RegisterReceiveAudioPayload(
     }
   }
   WebRtc_UWord8 bitsPerSample = 0; // zero implies frame based
+  bool isTrueStereo = false; // Default value
   if (ModuleRTPUtility::StringCompare(payloadName, "DVI4", 4)) {
     bitsPerSample = 4;
   } else if(ModuleRTPUtility::StringCompare(payloadName, "G722", 4)) {
@@ -250,6 +251,9 @@ ModuleRTPUtility::Payload* RTPReceiverAudio::RegisterReceiveAudioPayload(
     bitsPerSample = 8;
   } else if(ModuleRTPUtility::StringCompare(payloadName,"PCMA",4)) {
     bitsPerSample = 8;
+  } else if(ModuleRTPUtility::StringCompare(payloadName,"CELT",4))
+  {
+    isTrueStereo = true;
   }
   ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload;
   payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
@@ -258,6 +262,7 @@ ModuleRTPUtility::Payload* RTPReceiverAudio::RegisterReceiveAudioPayload(
   payload->typeSpecific.Audio.channels = channels;
   payload->typeSpecific.Audio.bitsPerSample = bitsPerSample;
   payload->typeSpecific.Audio.rate = rate;
+  payload->typeSpecific.Audio.trueStereoCodec = isTrueStereo;
   payload->audio = true;
   return payload;
 }
@@ -539,6 +544,18 @@ RTPReceiverAudio::ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
                                                            rtpHeader);
                 }
             }
+        } else if (audioSpecific.trueStereoCodec)
+        {
+            // One callback with the whole payload for each channel.
+            for(int channel = 1; (channel <= audioSpecific.channels) &&
+                (retVal == 0); channel++)
+            {
+                // One callback per channel.
+                rtpHeader->type.Audio.channel = channel;
+                retVal = CallbackOfReceivedPayloadData(payloadData,
+                                                       payloadLength,
+                                                       rtpHeader);
+            }
         } else
         {
             for(int channel = 1; channel <= audioSpecific.channels && retVal == 0; channel++)
diff --git a/src/modules/rtp_rtcp/source/rtp_utility.h b/src/modules/rtp_rtcp/source/rtp_utility.h
index 89012235f..70d7a8616 100644
--- a/src/modules/rtp_rtcp/source/rtp_utility.h
+++ b/src/modules/rtp_rtcp/source/rtp_utility.h
@@ -42,6 +42,7 @@ namespace ModuleRTPUtility
         WebRtc_UWord8     channels;
         WebRtc_UWord8     bitsPerSample;
         WebRtc_UWord32    rate;
+        bool              trueStereoCodec;
     };
     struct VideoPayload
     {