diff --git a/src/modules/audio_processing/main/source/audio_processing_impl.cc b/src/modules/audio_processing/main/source/audio_processing_impl.cc
index 9b4808be8..b1464e12a 100644
--- a/src/modules/audio_processing/main/source/audio_processing_impl.cc
+++ b/src/modules/audio_processing/main/source/audio_processing_impl.cc
@@ -256,8 +256,7 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
     return kNullPointerError;
   }
 
-  if (frame->_frequencyInHz !=
-      static_cast<WebRtc_UWord32>(sample_rate_hz_)) {
+  if (frame->_frequencyInHz != sample_rate_hz_) {
     return kBadSampleRateError;
   }
 
@@ -382,8 +381,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
     return kNullPointerError;
   }
 
-  if (frame->_frequencyInHz !=
-      static_cast<WebRtc_UWord32>(sample_rate_hz_)) {
+  if (frame->_frequencyInHz != sample_rate_hz_) {
     return kBadSampleRateError;
   }
 
diff --git a/src/modules/interface/module_common_types.h b/src/modules/interface/module_common_types.h
index 1865b0f60..3df370a47 100644
--- a/src/modules/interface/module_common_types.h
+++ b/src/modules/interface/module_common_types.h
@@ -697,7 +697,7 @@ public:
         const WebRtc_UWord32 timeStamp,
         const WebRtc_Word16* payloadData,
         const WebRtc_UWord16 payloadDataLengthInSamples,
-        const WebRtc_UWord32 frequencyInHz,
+        const int frequencyInHz,
         const SpeechType     speechType,
         const VADActivity    vadActivity,
         const WebRtc_UWord8  audioChannel = 1,
@@ -719,7 +719,7 @@ public:
     // Supporting Stereo, stereo samples are interleaved
     mutable WebRtc_Word16 _payloadData[kMaxAudioFrameSizeSamples];
     WebRtc_UWord16 _payloadDataLengthInSamples;
-    WebRtc_UWord32 _frequencyInHz;
+    int _frequencyInHz;
     WebRtc_UWord8  _audioChannel;
     SpeechType   _speechType;
     VADActivity  _vadActivity;
@@ -756,7 +756,7 @@ AudioFrame::UpdateFrame(
     const WebRtc_UWord32 timeStamp,
     const WebRtc_Word16* payloadData,
     const WebRtc_UWord16 payloadDataLengthInSamples,
-    const WebRtc_UWord32 frequencyInHz,
+    const int frequencyInHz,
     const SpeechType     speechType,
     const VADActivity    vadActivity,
     const WebRtc_UWord8  audioChannel,
diff --git a/src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
index e29a62d5a..28178fca8 100644
--- a/src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ b/src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
@@ -159,7 +159,7 @@ public:
     virtual WebRtc_Word32 OnInitializeDecoder(const WebRtc_Word32 id,
                                             const WebRtc_Word8 payloadType,
                                             const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
-                                            const WebRtc_UWord32 frequency,
+                                            const int frequency,
                                             const WebRtc_UWord8 channels,
                                             const WebRtc_UWord32 rate) = 0;
 
diff --git a/src/video_engine/main/source/vie_channel.cc b/src/video_engine/main/source/vie_channel.cc
index 5afcb17d8..9dbb99510 100644
--- a/src/video_engine/main/source/vie_channel.cc
+++ b/src/video_engine/main/source/vie_channel.cc
@@ -3142,7 +3142,7 @@ void ViEChannel::OnApplicationDataReceived(const WebRtc_Word32 id,
 WebRtc_Word32 ViEChannel::OnInitializeDecoder(
     const WebRtc_Word32 id, const WebRtc_Word8 payloadType,
     const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
-    const WebRtc_UWord32 frequency, const WebRtc_UWord8 channels,
+    const int frequency, const WebRtc_UWord8 channels,
     const WebRtc_UWord32 rate)
 {
     WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(_engineId, _channelId),
diff --git a/src/video_engine/main/source/vie_channel.h b/src/video_engine/main/source/vie_channel.h
index a4746b83b..5cce009ba 100644
--- a/src/video_engine/main/source/vie_channel.h
+++ b/src/video_engine/main/source/vie_channel.h
@@ -193,7 +193,7 @@ public:
         const WebRtc_Word32 id,
         const WebRtc_Word8 payloadType,
         const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
-        const WebRtc_UWord32 frequency,
+        const int frequency,
         const WebRtc_UWord8 channels,
         const WebRtc_UWord32 rate);
 
diff --git a/src/voice_engine/main/source/channel.cc b/src/voice_engine/main/source/channel.cc
index 190c2201a..78d552e6a 100644
--- a/src/voice_engine/main/source/channel.cc
+++ b/src/voice_engine/main/source/channel.cc
@@ -630,7 +630,7 @@ Channel::OnInitializeDecoder(
     const WebRtc_Word32 id,
     const WebRtc_Word8 payloadType,
     const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
-    const WebRtc_UWord32 frequency,
+    const int frequency,
     const WebRtc_UWord8 channels,
     const WebRtc_UWord32 rate)
 {
@@ -5860,7 +5860,7 @@ Channel::Demultiplex(const AudioFrame& audioFrame,
 }
 
 WebRtc_UWord32
-Channel::PrepareEncodeAndSend(WebRtc_UWord32 mixingFrequency)
+Channel::PrepareEncodeAndSend(int mixingFrequency)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::PrepareEncodeAndSend()");
@@ -6216,7 +6216,7 @@ Channel::GetRtpRtcp(RtpRtcp* &rtpRtcpModule) const
 }
 
 WebRtc_Word32
-Channel::MixOrReplaceAudioWithFile(const WebRtc_UWord32 mixingFrequency)
+Channel::MixOrReplaceAudioWithFile(const int mixingFrequency)
 {
     WebRtc_Word16 fileBuffer[320];
     WebRtc_UWord32 fileSamples(0);
@@ -6278,7 +6278,7 @@ Channel::MixOrReplaceAudioWithFile(const WebRtc_UWord32 mixingFrequency)
 
 WebRtc_Word32
 Channel::MixAudioWithFile(AudioFrame& audioFrame,
-                          const WebRtc_UWord32 mixingFrequency)
+                          const int mixingFrequency)
 {
     assert(mixingFrequency <= 32000);
 
diff --git a/src/voice_engine/main/source/channel.h b/src/voice_engine/main/source/channel.h
index 95cfd6341..7285c2f70 100644
--- a/src/voice_engine/main/source/channel.h
+++ b/src/voice_engine/main/source/channel.h
@@ -389,7 +389,7 @@ public:
             const WebRtc_Word32 id,
             const WebRtc_Word8 payloadType,
             const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
-            const WebRtc_UWord32 frequency,
+            const int frequency,
             const WebRtc_UWord8 channels,
             const WebRtc_UWord32 rate);
 
@@ -533,15 +533,15 @@ public:
 #endif
     WebRtc_UWord32 Demultiplex(const AudioFrame& audioFrame,
                                const WebRtc_UWord8 audioLevel_dBov);
-    WebRtc_UWord32 PrepareEncodeAndSend(WebRtc_UWord32 mixingFrequency);
+    WebRtc_UWord32 PrepareEncodeAndSend(int mixingFrequency);
     WebRtc_UWord32 EncodeAndSend();
 
 private:
     int InsertInbandDtmfTone();
     WebRtc_Word32
-            MixOrReplaceAudioWithFile(const WebRtc_UWord32 mixingFrequency);
+            MixOrReplaceAudioWithFile(const int mixingFrequency);
     WebRtc_Word32 MixAudioWithFile(AudioFrame& audioFrame,
-                                   const WebRtc_UWord32 mixingFrequency);
+                                   const int mixingFrequency);
     WebRtc_Word32 GetPlayoutTimeStamp(WebRtc_UWord32& playoutTimestamp);
     void UpdateDeadOrAliveCounters(bool alive);
     WebRtc_Word32 SendPacketRaw(const void *data, int len, bool RTCP);
@@ -576,9 +576,9 @@ private:
     FilePlayer* _inputFilePlayerPtr;
     FilePlayer* _outputFilePlayerPtr;
     FileRecorder* _outputFileRecorderPtr;
-    WebRtc_UWord32 _inputFilePlayerId;
-    WebRtc_UWord32 _outputFilePlayerId;
-    WebRtc_UWord32 _outputFileRecorderId;
+    int _inputFilePlayerId;
+    int _outputFilePlayerId;
+    int _outputFileRecorderId;
     bool _inputFilePlaying;
     bool _outputFilePlaying;
     bool _outputFileRecording;
diff --git a/src/voice_engine/main/source/output_mixer.h b/src/voice_engine/main/source/output_mixer.h
index 724cdf5b1..7fc1c4e32 100644
--- a/src/voice_engine/main/source/output_mixer.h
+++ b/src/voice_engine/main/source/output_mixer.h
@@ -142,12 +142,12 @@ private:  // owns
     Resampler _apmResampler;    // converts mixed audio to fit APM rate
     AudioLevel _audioLevel;    // measures audio level for the combined signal
     DtmfInband _dtmfGenerator;
-    WebRtc_UWord32 _instanceId;
+    int _instanceId;
     VoEMediaProcess* _externalMediaCallbackPtr;
     bool _externalMedia;
     float _panLeft;
     float _panRight;
-    WebRtc_UWord32 _mixingFrequencyHz;
+    int _mixingFrequencyHz;
     FileRecorder* _outputFileRecorderPtr;
     bool _outputFileRecording;
 };
diff --git a/src/voice_engine/main/source/transmit_mixer.cc b/src/voice_engine/main/source/transmit_mixer.cc
index 00c0d0188..570baa53b 100644
--- a/src/voice_engine/main/source/transmit_mixer.cc
+++ b/src/voice_engine/main/source/transmit_mixer.cc
@@ -314,7 +314,7 @@ TransmitMixer::PrepareDemux(const WebRtc_Word8* audioSamples,
                  totalDelayMS, clockDrift, currentMicLevel);
 
 
-    const WebRtc_UWord32 mixingFrequency = _mixingFrequency;
+    const int mixingFrequency = _mixingFrequency;
 
     ScopedChannel sc(*_channelManagerPtr);
     void* iterator(NULL);
@@ -326,7 +326,7 @@ TransmitMixer::PrepareDemux(const WebRtc_Word8* audioSamples,
         {
             CodecInst tmpCdc;
             channelPtr->GetSendCodec(tmpCdc);
-            if ((WebRtc_UWord32) tmpCdc.plfreq > _mixingFrequency)
+            if (tmpCdc.plfreq > _mixingFrequency)
                 _mixingFrequency = tmpCdc.plfreq;
         }
         channelPtr = sc.GetNextChannel(iterator);
@@ -1151,7 +1151,7 @@ TransmitMixer::GenerateAudioFrame(const WebRtc_Word16 audioSamples[],
                                   const WebRtc_UWord32 nSamples,
                                   const WebRtc_UWord8 nChannels,
                                   const WebRtc_UWord32 samplesPerSec,
-                                  const WebRtc_UWord32 mixingFrequency)
+                                  const int mixingFrequency)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::GenerateAudioFrame(nSamples=%u,"
@@ -1214,12 +1214,11 @@ WebRtc_Word32 TransmitMixer::RecordAudioToFile(
 }
 
 WebRtc_Word32 TransmitMixer::MixOrReplaceAudioWithFile(
-    const WebRtc_UWord32 mixingFrequency)
+    const int mixingFrequency)
 {
     WebRtc_Word16 fileBuffer[320];
 
     WebRtc_UWord32 fileSamples(0);
-    WebRtc_Word32 outSamples(0);
 
     {
         CriticalSectionScoped cs(_critSect);
@@ -1387,7 +1386,6 @@ int TransmitMixer::TypingDetection()
     {
         return (-1);
     }
-    bool vad = (_audioFrame._vadActivity == AudioFrame::kVadActive);
 
     if (_audioFrame._vadActivity == AudioFrame::kVadActive)
         _timeActive++;
@@ -1423,7 +1421,7 @@ int TransmitMixer::TypingDetection()
 }
 #endif
 
-WebRtc_UWord32 TransmitMixer::GetMixingFrequency()
+int TransmitMixer::GetMixingFrequency()
 {
     assert(_mixingFrequency!=0);
     return (_mixingFrequency);
diff --git a/src/voice_engine/main/source/transmit_mixer.h b/src/voice_engine/main/source/transmit_mixer.h
index 871521e9c..b832b8b9c 100644
--- a/src/voice_engine/main/source/transmit_mixer.h
+++ b/src/voice_engine/main/source/transmit_mixer.h
@@ -81,7 +81,7 @@ public:
 
     int DeRegisterExternalMediaProcessing();
 
-    WebRtc_UWord32 GetMixingFrequency();
+    int GetMixingFrequency();
 
     // VoEVolumeControl
     int SetMute(const bool enable);
@@ -160,11 +160,11 @@ private:
                                      const WebRtc_UWord32 nSamples,
                                      const WebRtc_UWord8 nChannels,
                                      const WebRtc_UWord32 samplesPerSec,
-                                     const WebRtc_UWord32 mixingFrequency);
+                                     const int mixingFrequency);
     WebRtc_Word32 RecordAudioToFile(const WebRtc_UWord32 mixingFrequency);
 
     WebRtc_Word32 MixOrReplaceAudioWithFile(
-        const WebRtc_UWord32 mixingFrequency);
+        const int mixingFrequency);
 
     WebRtc_Word32 APMProcessStream(const WebRtc_UWord16 totalDelayMS,
                                    const WebRtc_Word32 clockDrift,
@@ -188,9 +188,9 @@ private:  // owns
     FilePlayer*	_filePlayerPtr;
     FileRecorder* _fileRecorderPtr;
     FileRecorder* _fileCallRecorderPtr;
-    WebRtc_UWord32 _filePlayerId;
-    WebRtc_UWord32 _fileRecorderId;
-    WebRtc_UWord32 _fileCallRecorderId;
+    int _filePlayerId;
+    int _fileRecorderId;
+    int _fileCallRecorderId;
     bool _filePlaying;
     bool _fileRecording;
     bool _fileCallRecording;
@@ -208,14 +208,14 @@ private:  // owns
     WebRtc_UWord32 _noiseWarning;
 
 private:
-    WebRtc_UWord32 _instanceId;
+    int _instanceId;
     bool _mixFileWithMicrophone;
     WebRtc_UWord32 _captureLevel;
     bool _externalMedia;
     VoEMediaProcess* _externalMediaCallbackPtr;
     bool _mute;
     WebRtc_Word32 _remainingMuteMicTimeMs;
-    WebRtc_UWord32 _mixingFrequency;
+    int _mixingFrequency;
     bool _includeAudioLevelIndication;
     WebRtc_UWord8 _audioLevel_dBov;
 };
diff --git a/src/voice_engine/main/source/voe_base_impl.cc b/src/voice_engine/main/source/voe_base_impl.cc
index 478f0b7c7..a25a3e4ca 100644
--- a/src/voice_engine/main/source/voe_base_impl.cc
+++ b/src/voice_engine/main/source/voe_base_impl.cc
@@ -282,7 +282,8 @@ WebRtc_Word32 VoEBaseImpl::NeedMorePlayData(
     _outputMixerPtr->GetMixedAudio(samplesPerSec, nChannels, audioFrame);
 
     assert(nSamples == audioFrame._payloadDataLengthInSamples);
-    assert(samplesPerSec == audioFrame._frequencyInHz);
+    assert(samplesPerSec ==
+        static_cast<WebRtc_UWord32>(audioFrame._frequencyInHz));
 
     // Deliver audio (PCM) samples to the ADM
     memcpy(
@@ -585,7 +586,6 @@ int VoEBaseImpl::Init()
     }
 
     bool available(false);
-    WebRtc_Word32 ret(0);
 
     // --------------------
     // Reinitialize the ADM
diff --git a/src/voice_engine/main/source/voe_call_report_impl.cc b/src/voice_engine/main/source/voe_call_report_impl.cc
index 20c32e57d..9ae4f320a 100644
--- a/src/voice_engine/main/source/voe_call_report_impl.cc
+++ b/src/voice_engine/main/source/voe_call_report_impl.cc
@@ -85,8 +85,6 @@ int VoECallReportImpl::ResetCallReportStatistics(int channel)
     }
     assert(_audioProcessingModulePtr != NULL);
 
-    int res1(0);
-    int res2(0);
     bool levelMode =
         _audioProcessingModulePtr->level_estimator()->is_enabled();
     bool echoMode =