diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h b/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
index 697121e50..5068c7f45 100644
--- a/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
+++ b/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
@@ -62,6 +62,13 @@ int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate);
 int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels);
 int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst);
 
+/****************************************************************************
+ * WebRtcOpus_DecoderChannels(...)
+ *
+ * This function returns the number of channels created for Opus decoder.
+ */
+int WebRtcOpus_DecoderChannels(OpusDecInst* inst);
+
 /****************************************************************************
  * WebRtcOpus_DecoderInit(...)
  *
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index 867262d0b..7da5df8a7 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
@@ -55,6 +55,7 @@ int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst, int32_t channels) {
 
 int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
   opus_encoder_destroy(inst->encoder);
+  free(inst);
   return 0;
 }
 
@@ -90,23 +91,36 @@ struct WebRtcOpusDecInst {
 };
 
 int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) {
+  int error_l;
+  int error_r;
   OpusDecInst* state;
+
+  // Create Opus decoder memory.
   state = (OpusDecInst*) calloc(1, sizeof(OpusDecInst));
-  if (state) {
-    int error_l;
-    int error_r;
-    // Always create a 48000 Hz Opus decoder.
-    state->decoder_left = opus_decoder_create(48000, channels, &error_l);
-    state->decoder_right = opus_decoder_create(48000, channels, &error_r);
-    if (error_l == OPUS_OK && error_r == OPUS_OK &&
-        state->decoder_left != NULL && state->decoder_right != NULL) {
-      state->channels = channels;
-      *inst = state;
-      return 0;
-    }
-    free(state);
-    state = NULL;
+  if (state == NULL) {
+    return -1;
   }
+
+  // Create new memory for left and right channel, always at 48000 Hz.
+  state->decoder_left = opus_decoder_create(48000, channels, &error_l);
+  state->decoder_right = opus_decoder_create(48000, channels, &error_r);
+  if (error_l == OPUS_OK && error_r == OPUS_OK && state->decoder_left != NULL
+      && state->decoder_right != NULL) {
+    // Creation of memory all ok.
+    state->channels = channels;
+    *inst = state;
+    return 0;
+  }
+
+  // If memory allocation was unsuccessful, free the entire state.
+  if (state->decoder_left) {
+    opus_decoder_destroy(state->decoder_left);
+  }
+  if (state->decoder_right) {
+    opus_decoder_destroy(state->decoder_right);
+  }
+  free(state);
+  state = NULL;
   return -1;
 }
 
@@ -117,6 +131,10 @@ int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst) {
   return 0;
 }
 
+int WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
+  return inst->channels;
+}
+
 int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst) {
   int error = opus_decoder_ctl(inst->decoder_left, OPUS_RESET_STATE);
   if (error == OPUS_OK) {
diff --git a/webrtc/modules/audio_coding/main/source/acm_codec_database.cc b/webrtc/modules/audio_coding/main/source/acm_codec_database.cc
index 338dce139..19bccb649 100644
--- a/webrtc/modules/audio_coding/main/source/acm_codec_database.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_codec_database.cc
@@ -190,10 +190,8 @@ const CodecInst ACMCodecDB::database_[] = {
 #endif
 #ifdef WEBRTC_CODEC_OPUS
   // Opus internally supports 48, 24, 16, 12, 8 kHz.
-  // Mono
-  {120, "opus", 48000, 960, 1, 32000},
-  // Stereo
-  {121, "opus", 48000, 960, 2, 32000},
+  // Mono and stereo.
+  {120, "opus", 48000, 960, 2, 32000},
 #endif
 #ifdef WEBRTC_CODEC_SPEEX
   {kDynamicPayloadtypes[count_database++], "speex", 8000, 160, 1, 11000},
@@ -285,9 +283,7 @@ const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
 #ifdef WEBRTC_CODEC_OPUS
   // Opus supports frames shorter than 10ms,
   // but it doesn't help us to use them.
-  // Mono
-  {1, {960}, 0, 2},
-  // Stereo
+  // Mono and stereo.
   {1, {960}, 0, 2},
 #endif
 #ifdef WEBRTC_CODEC_SPEEX
@@ -375,10 +371,8 @@ const WebRtcNetEQDecoder ACMCodecDB::neteq_decoders_[] = {
   kDecoderGSMFR,
 #endif
 #ifdef WEBRTC_CODEC_OPUS
-  // Mono
+  // Mono and stereo.
   kDecoderOpus,
-  // Stereo
-  kDecoderOpus_2ch,
 #endif
 #ifdef WEBRTC_CODEC_SPEEX
   kDecoderSPEEX_8,
@@ -571,7 +565,13 @@ int ACMCodecDB::CodecId(const char* payload_name, int frequency, int channels) {
     // always treated as true, like for RED.
     name_match = (STR_CASE_CMP(database_[id].plname, payload_name) == 0);
     frequency_match = (frequency == database_[id].plfreq) || (frequency == -1);
-    channels_match = (channels == database_[id].channels);
+    // The number of channels must match for all codecs but Opus.
+    if (STR_CASE_CMP(payload_name, "opus") != 0) {
+      channels_match = (channels == database_[id].channels);
+    } else {
+      // For opus we just check that number of channels is valid.
+      channels_match = (channels == 1 || channels == 2);
+    }
 
     if (name_match && frequency_match && channels_match) {
       // We have found a matching codec in the list.
@@ -767,11 +767,7 @@ ACMGenericCodec* ACMCodecDB::CreateCodecInstance(const CodecInst* codec_inst) {
 #endif
   } else if (!STR_CASE_CMP(codec_inst->plname, "opus")) {
 #ifdef WEBRTC_CODEC_OPUS
-    if (codec_inst->channels == 1) {
-      return new ACMOpus(kOpus);
-    } else {
-      return new ACMOpus(kOpus_2ch);
-    }
+    return new ACMOpus(kOpus);
 #endif
   } else if (!STR_CASE_CMP(codec_inst->plname, "speex")) {
 #ifdef WEBRTC_CODEC_SPEEX
diff --git a/webrtc/modules/audio_coding/main/source/acm_codec_database.h b/webrtc/modules/audio_coding/main/source/acm_codec_database.h
index 60578db0b..482215395 100644
--- a/webrtc/modules/audio_coding/main/source/acm_codec_database.h
+++ b/webrtc/modules/audio_coding/main/source/acm_codec_database.h
@@ -92,10 +92,8 @@ class ACMCodecDB {
     , kGSMFR
 #endif
 #ifdef WEBRTC_CODEC_OPUS
-    // Mono
+    // Mono and stereo
     , kOpus
-    // Stereo
-    , kOpus_2ch
 #endif
 #ifdef WEBRTC_CODEC_SPEEX
     , kSPEEX8
@@ -178,10 +176,8 @@ class ACMCodecDB {
   enum {kSPEEX16 = -1};
 #endif
 #ifndef WEBRTC_CODEC_OPUS
-  // Mono
+  // Mono and stereo
   enum {kOpus = -1};
-  // Stereo
-  enum {kOpus_2ch = -1};
 #endif
 #ifndef WEBRTC_CODEC_AVT
   enum {kAVT = -1};
diff --git a/webrtc/modules/audio_coding/main/source/acm_opus.cc b/webrtc/modules/audio_coding/main/source/acm_opus.cc
index ca2892161..4e9e5a25c 100644
--- a/webrtc/modules/audio_coding/main/source/acm_opus.cc
+++ b/webrtc/modules/audio_coding/main/source/acm_opus.cc
@@ -111,7 +111,7 @@ ACMOpus::ACMOpus(int16_t codecID)
   // Opus has internal DTX, but we dont use it for now.
   _hasInternalDTX = false;
 
-  if ((_codecID != ACMCodecDB::kOpus) && (_codecID != ACMCodecDB::kOpus_2ch)) {
+  if (_codecID != ACMCodecDB::kOpus) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
                  "Wrong codec id for Opus.");
     _sampleFreq = -1;
@@ -190,15 +190,17 @@ int16_t ACMOpus::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
 }
 
 int16_t ACMOpus::InternalInitDecoder(WebRtcACMCodecParams* codecParams) {
-  if (_decoderInstPtr != NULL) {
-    WebRtcOpus_DecoderFree(_decoderInstPtr);
-    _decoderInstPtr = NULL;
-  }
-  if (WebRtcOpus_DecoderCreate(&_decoderInstPtr,
-                               codecParams->codecInstant.channels) < 0) {
-    return -1;
+  if(_decoderInstPtr == NULL) {
+    if (WebRtcOpus_DecoderCreate(&_decoderInstPtr,
+                                 codecParams->codecInstant.channels) < 0) {
+      return -1;
+    }
   }
 
+  // Number of channels in decoder should match the number in |codecParams|.
+  assert(codecParams->codecInstant.channels ==
+      WebRtcOpus_DecoderChannels(_decoderInstPtr));
+
   if (WebRtcOpus_DecoderInit(_decoderInstPtr) < 0) {
     return -1;
   }
@@ -221,13 +223,8 @@ int32_t ACMOpus::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
   // TODO(tlegrand): Decoder is registered in NetEQ as a 32 kHz decoder, which
   // is true until we have a full 48 kHz system, and remove the downsampling
   // in the Opus decoder wrapper.
-  if (codecInst.channels == 1) {
-    SET_CODEC_PAR(codecDef, kDecoderOpus, codecInst.pltype, _decoderInstPtr,
-                  32000);
-  } else {
-    SET_CODEC_PAR(codecDef, kDecoderOpus_2ch, codecInst.pltype,
-                  _decoderInstPtr, 32000);
-  }
+  SET_CODEC_PAR(codecDef, kDecoderOpus, codecInst.pltype,
+                _decoderInstPtr, 32000);
 
   // If this is the master of NetEQ, regular decoder will be added, otherwise
   // the slave decoder will be used.
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module.cc
index 0efc8b7c1..12ebf00db 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module.cc
@@ -59,6 +59,10 @@ WebRtc_Word32 AudioCodingModule::Codec(const char* payload_name,
   // Get default codec settings.
   ACMCodecDB::Codec(codec_id, &codec);
 
+  // Keep the number of channels from the function call. For most codecs it
+  // will be the same value as in defaul codec settings, but not for all.
+  codec.channels = channels;
+
   return 0;
 }
 
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
index 0a399ba3f..f586c224b 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
@@ -1285,7 +1285,7 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutFrequency() const {
   return _netEq.CurrentSampFreqHz();
 }
 
-// Register possible reveive codecs, can be called multiple times,
+// Register possible receive codecs, can be called multiple times,
 // for codecs, CNG (NB, WB and SWB), DTMF, RED.
 WebRtc_Word32 AudioCodingModuleImpl::RegisterReceiveCodec(
     const CodecInst& receive_codec) {
@@ -1394,14 +1394,30 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterReceiveCodec(
 
     if (!_stereoReceive[codec_id]
         && (_lastRecvAudioCodecPlType == receive_codec.pltype)) {
-      // The last received payload type is the same as the current one, but
-      // was marked as mono. Reset to avoid problems.
+      // The last received payload type is the same as the one we are
+      // registering. Expected number of channels to receive is one (mono),
+      // but we are now registering the receiving codec as stereo (number of
+      // channels is 2).
+      // Set |_lastRecvAudioCodedPlType| to invalid value to trigger a flush in
+      // NetEq, and a reset of expected number of channels next time a packet is
+      // received in AudioCodingModuleImpl::IncomingPacket().
       _lastRecvAudioCodecPlType = -1;
     }
 
     _stereoReceive[codec_id] = true;
     _stereoReceiveRegistered = true;
   } else {
+    if (_lastRecvAudioCodecPlType == receive_codec.pltype &&
+        _expected_channels == 2) {
+      // The last received payload type is the same as the one we are
+      // registering. Expected number of channels to receive is two (stereo),
+      // but we are now registering the receiving codec as mono (number of
+      // channels is 1).
+      // Set |_lastRecvAudioCodedPlType| to invalid value to trigger a flush in
+      // NetEq, and a reset of expected number of channels next time a packet is
+      // received in AudioCodingModuleImpl::IncomingPacket().
+      _lastRecvAudioCodecPlType = -1;
+    }
     _stereoReceive[codec_id] = false;
   }
 
diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
index 89a9829a8..d4d74c215 100644
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
+++ b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
@@ -147,6 +147,9 @@ void TestAllCodecs::Perform() {
   CodecInst my_codec_param;
   for (uint8_t n = 0; n < num_encoders; n++) {
     acm_b_->Codec(n, my_codec_param);
+    if (!strcmp(my_codec_param.plname, "opus")) {
+      my_codec_param.channels = 1;
+    }
     acm_b_->RegisterReceiveCodec(my_codec_param);
   }
 
@@ -635,6 +638,7 @@ void TestAllCodecs::Perform() {
   Run(channel_a_to_b_);
   RegisterSendCodec('A', codec_opus, 48000, 500000, 960, -1);
   Run(channel_a_to_b_);
+  outfile_b_.Close();
 #endif
   if (test_mode_ != 0) {
     printf("===============================================================\n");
diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.cc b/webrtc/modules/audio_coding/main/test/TestStereo.cc
index 97d6c2ebd..3a4b65237 100644
--- a/webrtc/modules/audio_coding/main/test/TestStereo.cc
+++ b/webrtc/modules/audio_coding/main/test/TestStereo.cc
@@ -550,12 +550,18 @@ void TestStereo::Perform() {
     printf("Test number: %d\n",test_cntr_ + 1);
     printf("Test type: Mono-to-stereo\n");
   }
+
+  // Keep encode and decode in stereo.
   test_cntr_++;
   channel_a2b_->set_codec_mode(kStereo);
   OpenOutFile(test_cntr_);
   RegisterSendCodec('A', codec_opus, 48000, 64000, 960, codec_channels,
                     opus_pltype_);
   Run(channel_a2b_, audio_channels, codec_channels);
+
+  // Encode in mono, decode in stereo mode.
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 960, 1, opus_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
   out_file_.Close();
 #endif
 
@@ -661,10 +667,63 @@ void TestStereo::Perform() {
   }
   test_cntr_++;
   OpenOutFile(test_cntr_);
+  // Encode and decode in mono.
   RegisterSendCodec('A', codec_opus, 48000, 32000, 960, codec_channels,
                      opus_pltype_);
+  CodecInst opus_codec_param;
+  for (WebRtc_UWord8 n = 0; n < num_encoders; n++) {
+    EXPECT_EQ(0, acm_b_->Codec(n, opus_codec_param));
+    if (!strcmp(opus_codec_param.plname, "opus")) {
+      opus_codec_param.channels = 1;
+      EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(opus_codec_param));
+      break;
+    }
+  }
+  Run(channel_a2b_, audio_channels, codec_channels);
+
+  // Encode in stereo, decode in mono.
+  RegisterSendCodec('A', codec_opus, 48000, 32000, 960, 2, opus_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+
+  out_file_.Close();
+
+  // Test switching between decoding mono and stereo for Opus.
+
+  // Decode in mono.
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  if (test_mode_ != 0) {
+    // Print out codec and settings
+    printf("Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
+        " Decode: mono\n", test_cntr_);
+  }
   Run(channel_a2b_, audio_channels, codec_channels);
   out_file_.Close();
+  // Decode in stereo.
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  if (test_mode_ != 0) {
+    // Print out codec and settings
+    printf("Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
+        " Decode: stereo\n", test_cntr_);
+  }
+  opus_codec_param.channels = 2;
+  EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(opus_codec_param));
+  Run(channel_a2b_, audio_channels, 2);
+  out_file_.Close();
+  // Decode in mono.
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  if (test_mode_ != 0) {
+    // Print out codec and settings
+    printf("Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
+        " Decode: mono\n", test_cntr_);
+  }
+  opus_codec_param.channels = 1;
+  EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(opus_codec_param));
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+
 #endif
 
   // Print out which codecs were tested, and which were not, in the run.
@@ -679,6 +738,9 @@ void TestStereo::Perform() {
     printf("   G.711\n");
 #ifdef WEBRTC_CODEC_CELT
     printf("   CELT\n");
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+    printf("   Opus\n");
 #endif
     printf("\nTo complete the test, listen to the %d number of output "
            "files.\n",
diff --git a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc b/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
index fcca3748d..567903b26 100644
--- a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
+++ b/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
@@ -83,6 +83,10 @@ void TestVADDTX::Perform()
         {
             printf("%s\n", myCodecParam.plname);
         }
+        if (!strcmp(myCodecParam.plname, "opus")) {
+          // Use mono decoding for Opus in the VAD/DTX test.
+          myCodecParam.channels = 1;
+        }
         _acmB->RegisterReceiveCodec(myCodecParam);
     }
 
@@ -337,6 +341,8 @@ WebRtc_Word16 TestVADDTX::RegisterSendCodec(char side,
         }
     }
 
+    // We only allow VAD/DTX when sending mono.
+    myCodecParam.channels = 1;
     CHECK_ERROR(myACM->RegisterSendCodec(myCodecParam));
 
     // initialization was succesful
diff --git a/webrtc/modules/audio_coding/neteq/codec_db.c b/webrtc/modules/audio_coding/neteq/codec_db.c
index fd5988699..10277d506 100644
--- a/webrtc/modules/audio_coding/neteq/codec_db.c
+++ b/webrtc/modules/audio_coding/neteq/codec_db.c
@@ -117,7 +117,6 @@ int WebRtcNetEQ_DbAdd(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec,
 #endif
 #ifdef NETEQ_OPUS_CODEC
         case kDecoderOpus :
-        case kDecoderOpus_2ch :
 #endif
 #ifdef NETEQ_G722_CODEC
         case kDecoderG722 :
@@ -466,7 +465,6 @@ int WebRtcNetEQ_DbGetSplitInfo(SplitInfo_t *inst, enum WebRtcNetEQDecoder codecI
 #endif
 #ifdef NETEQ_OPUS_CODEC
         case kDecoderOpus:
-        case kDecoderOpus_2ch :
 #endif
 #ifdef NETEQ_ARBITRARY_CODEC
         case kDecoderArbitrary:
diff --git a/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h b/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h
index 5af201e04..9621036dc 100644
--- a/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h
+++ b/webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h
@@ -63,7 +63,6 @@ enum WebRtcNetEQDecoder
     kDecoderG722_1C_32,
     kDecoderG722_1C_48,
     kDecoderOpus,
-    kDecoderOpus_2ch,
     kDecoderSPEEX_8,
     kDecoderSPEEX_16,
     kDecoderCELT_32,
diff --git a/webrtc/modules/audio_coding/neteq/packet_buffer.c b/webrtc/modules/audio_coding/neteq/packet_buffer.c
index e8ee40c91..c4c301cd0 100644
--- a/webrtc/modules/audio_coding/neteq/packet_buffer.c
+++ b/webrtc/modules/audio_coding/neteq/packet_buffer.c
@@ -621,8 +621,7 @@ int WebRtcNetEQ_GetDefaultCodecSettings(const enum WebRtcNetEQDecoder *codecID,
             codecBytes = 1560; /* 240ms @ 52kbps (30ms frames) */
             codecBuffers = 8;
         }
-        else if ((codecID[i] == kDecoderOpus)  ||
-            (codecID[i] == kDecoderOpus_2ch))
+        else if (codecID[i] == kDecoderOpus)
         {
             codecBytes = 15300; /* 240ms @ 510kbps (60ms frames) */
             codecBuffers = 30;  /* Replicating the value for PCMu/a */
diff --git a/webrtc/modules/audio_coding/neteq/recin.c b/webrtc/modules/audio_coding/neteq/recin.c
index 24e1eeed2..00c8f81da 100644
--- a/webrtc/modules/audio_coding/neteq/recin.c
+++ b/webrtc/modules/audio_coding/neteq/recin.c
@@ -381,7 +381,6 @@ int WebRtcNetEQ_GetTimestampScaling(MCUInst_t *MCU_inst, int rtpPayloadType)
             break;
         }
         case kDecoderOpus:
-        case kDecoderOpus_2ch:
         {
             /* We resample Opus internally to 32 kHz, but timestamps
              * are counted at 48 kHz. So there are two output samples