diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index d6629029b..0728990c3 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -94,7 +94,7 @@ Channel::InFrameType(int16_t frameType)
 }
 
 int32_t
-Channel::OnRxVadDetected(const int vadDecision)
+Channel::OnRxVadDetected(int vadDecision)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::OnRxVadDetected(vadDecision=%d)", vadDecision);
@@ -332,10 +332,10 @@ Channel::SendRTCPPacket(int channel, const void *data, int len)
 }
 
 void
-Channel::OnPlayTelephoneEvent(const int32_t id,
-                              const uint8_t event,
-                              const uint16_t lengthMs,
-                              const uint8_t volume)
+Channel::OnPlayTelephoneEvent(int32_t id,
+                              uint8_t event,
+                              uint16_t lengthMs,
+                              uint8_t volume)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::OnPlayTelephoneEvent(id=%d, event=%u, lengthMs=%u,"
@@ -356,8 +356,8 @@ Channel::OnPlayTelephoneEvent(const int32_t id,
 }
 
 void
-Channel::OnIncomingSSRCChanged(const int32_t id,
-                               const uint32_t SSRC)
+Channel::OnIncomingSSRCChanged(int32_t id,
+                               uint32_t SSRC)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::OnIncomingSSRCChanged(id=%d, SSRC=%d)",
@@ -382,9 +382,9 @@ Channel::OnIncomingSSRCChanged(const int32_t id,
     }
 }
 
-void Channel::OnIncomingCSRCChanged(const int32_t id,
-                                    const uint32_t CSRC,
-                                    const bool added)
+void Channel::OnIncomingCSRCChanged(int32_t id,
+                                    uint32_t CSRC,
+                                    bool added)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::OnIncomingCSRCChanged(id=%d, CSRC=%d, added=%d)",
@@ -405,10 +405,10 @@ void Channel::OnIncomingCSRCChanged(const int32_t id,
 }
 
 void
-Channel::OnApplicationDataReceived(const int32_t id,
-                                   const uint8_t subType,
-                                   const uint32_t name,
-                                   const uint16_t length,
+Channel::OnApplicationDataReceived(int32_t id,
+                                   uint8_t subType,
+                                   uint32_t name,
+                                   uint16_t length,
                                    const uint8_t* data)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
@@ -436,12 +436,12 @@ Channel::OnApplicationDataReceived(const int32_t id,
 
 int32_t
 Channel::OnInitializeDecoder(
-    const int32_t id,
-    const int8_t payloadType,
+    int32_t id,
+    int8_t payloadType,
     const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-    const int frequency,
-    const uint8_t channels,
-    const uint32_t rate)
+    int frequency,
+    uint8_t channels,
+    uint32_t rate)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::OnInitializeDecoder(id=%d, payloadType=%d, "
@@ -477,7 +477,7 @@ Channel::OnInitializeDecoder(
 }
 
 void
-Channel::OnPacketTimeout(const int32_t id)
+Channel::OnPacketTimeout(int32_t id)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::OnPacketTimeout(id=%d)", id);
@@ -504,8 +504,8 @@ Channel::OnPacketTimeout(const int32_t id)
 }
 
 void
-Channel::OnReceivedPacket(const int32_t id,
-                          const RtpRtcpPacketType packetType)
+Channel::OnReceivedPacket(int32_t id,
+                          RtpRtcpPacketType packetType)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::OnReceivedPacket(id=%d, packetType=%d)",
@@ -536,8 +536,8 @@ Channel::OnReceivedPacket(const int32_t id,
 }
 
 void
-Channel::OnPeriodicDeadOrAlive(const int32_t id,
-                               const RTPAliveType alive)
+Channel::OnPeriodicDeadOrAlive(int32_t id,
+                               RTPAliveType alive)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::OnPeriodicDeadOrAlive(id=%d, alive=%d)", id, alive);
@@ -590,7 +590,7 @@ Channel::OnPeriodicDeadOrAlive(const int32_t id,
 
 int32_t
 Channel::OnReceivedPayloadData(const uint8_t* payloadData,
-                               const uint16_t payloadSize,
+                               uint16_t payloadSize,
                                const WebRtcRTPHeader* rtpHeader)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
@@ -631,7 +631,7 @@ Channel::OnReceivedPayloadData(const uint8_t* payloadData,
     return 0;
 }
 
-int32_t Channel::GetAudioFrame(const int32_t id, AudioFrame& audioFrame)
+int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::GetAudioFrame(id=%d)", id);
@@ -737,7 +737,7 @@ int32_t Channel::GetAudioFrame(const int32_t id, AudioFrame& audioFrame)
 }
 
 int32_t
-Channel::NeededFrequency(const int32_t id)
+Channel::NeededFrequency(int32_t id)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::NeededFrequency(id=%d)", id);
@@ -778,8 +778,8 @@ Channel::NeededFrequency(const int32_t id)
 
 int32_t
 Channel::CreateChannel(Channel*& channel,
-                       const int32_t channelId,
-                       const uint32_t instanceId)
+                       int32_t channelId,
+                       uint32_t instanceId)
 {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId,channelId),
                  "Channel::CreateChannel(channelId=%d, instanceId=%d)",
@@ -798,7 +798,7 @@ Channel::CreateChannel(Channel*& channel,
 }
 
 void
-Channel::PlayNotification(const int32_t id, const uint32_t durationMs)
+Channel::PlayNotification(int32_t id, uint32_t durationMs)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::PlayNotification(id=%d, durationMs=%d)",
@@ -808,7 +808,7 @@ Channel::PlayNotification(const int32_t id, const uint32_t durationMs)
 }
 
 void
-Channel::RecordNotification(const int32_t id, const uint32_t durationMs)
+Channel::RecordNotification(int32_t id, uint32_t durationMs)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::RecordNotification(id=%d, durationMs=%d)",
@@ -818,7 +818,7 @@ Channel::RecordNotification(const int32_t id, const uint32_t durationMs)
 }
 
 void
-Channel::PlayFileEnded(const int32_t id)
+Channel::PlayFileEnded(int32_t id)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::PlayFileEnded(id=%d)", id);
@@ -846,7 +846,7 @@ Channel::PlayFileEnded(const int32_t id)
 }
 
 void
-Channel::RecordFileEnded(const int32_t id)
+Channel::RecordFileEnded(int32_t id)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::RecordFileEnded(id=%d)", id);
@@ -862,8 +862,8 @@ Channel::RecordFileEnded(const int32_t id)
                  " shutdown");
 }
 
-Channel::Channel(const int32_t channelId,
-                 const uint32_t instanceId) :
+Channel::Channel(int32_t channelId,
+                 uint32_t instanceId) :
     _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _instanceId(instanceId),
@@ -2300,11 +2300,11 @@ Channel::GetPeriodicDeadOrAliveStatus(bool& enabled, int& sampleTimeSeconds)
 }
 
 int Channel::StartPlayingFileLocally(const char* fileName,
-                                     const bool loop,
-                                     const FileFormats format,
-                                     const int startPosition,
-                                     const float volumeScaling,
-                                     const int stopPosition,
+                                     bool loop,
+                                     FileFormats format,
+                                     int startPosition,
+                                     float volumeScaling,
+                                     int stopPosition,
                                      const CodecInst* codecInst)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
@@ -2372,10 +2372,10 @@ int Channel::StartPlayingFileLocally(const char* fileName,
 }
 
 int Channel::StartPlayingFileLocally(InStream* stream,
-                                     const FileFormats format,
-                                     const int startPosition,
-                                     const float volumeScaling,
-                                     const int stopPosition,
+                                     FileFormats format,
+                                     int startPosition,
+                                     float volumeScaling,
+                                     int stopPosition,
                                      const CodecInst* codecInst)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
@@ -2530,7 +2530,7 @@ int Channel::RegisterFilePlayingToMixer()
     return 0;
 }
 
-int Channel::ScaleLocalFilePlayout(const float scale)
+int Channel::ScaleLocalFilePlayout(float scale)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::ScaleLocalFilePlayout(scale=%5.3f)", scale);
@@ -2586,11 +2586,11 @@ int Channel::GetLocalPlayoutPosition(int& positionMs)
 }
 
 int Channel::StartPlayingFileAsMicrophone(const char* fileName,
-                                          const bool loop,
-                                          const FileFormats format,
-                                          const int startPosition,
-                                          const float volumeScaling,
-                                          const int stopPosition,
+                                          bool loop,
+                                          FileFormats format,
+                                          int startPosition,
+                                          float volumeScaling,
+                                          int stopPosition,
                                           const CodecInst* codecInst)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
@@ -2655,10 +2655,10 @@ int Channel::StartPlayingFileAsMicrophone(const char* fileName,
 }
 
 int Channel::StartPlayingFileAsMicrophone(InStream* stream,
-                                          const FileFormats format,
-                                          const int startPosition,
-                                          const float volumeScaling,
-                                          const int stopPosition,
+                                          FileFormats format,
+                                          int startPosition,
+                                          float volumeScaling,
+                                          int stopPosition,
                                           const CodecInst* codecInst)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
@@ -2762,7 +2762,7 @@ int Channel::IsPlayingFileAsMicrophone() const
     return _inputFilePlaying;
 }
 
-int Channel::ScaleFileAsMicrophonePlayout(const float scale)
+int Channel::ScaleFileAsMicrophonePlayout(float scale)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::ScaleFileAsMicrophonePlayout(scale=%5.3f)", scale);
@@ -3280,7 +3280,7 @@ Channel::VoiceActivityIndicator(int &activity)
 #ifdef WEBRTC_VOICE_ENGINE_AGC
 
 int
-Channel::SetRxAgcStatus(const bool enable, const AgcModes mode)
+Channel::SetRxAgcStatus(bool enable, AgcModes mode)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::SetRxAgcStatus(enable=%d, mode=%d)",
@@ -3360,7 +3360,7 @@ Channel::GetRxAgcStatus(bool& enabled, AgcModes& mode)
 }
 
 int
-Channel::SetRxAgcConfig(const AgcConfig config)
+Channel::SetRxAgcConfig(AgcConfig config)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::SetRxAgcConfig()");
@@ -3424,7 +3424,7 @@ Channel::GetRxAgcConfig(AgcConfig& config)
 #ifdef WEBRTC_VOICE_ENGINE_NR
 
 int
-Channel::SetRxNsStatus(const bool enable, const NsModes mode)
+Channel::SetRxNsStatus(bool enable, NsModes mode)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::SetRxNsStatus(enable=%d, mode=%d)",
@@ -3888,7 +3888,7 @@ Channel::GetRemoteRTCPData(
 }
 
 int
-Channel::SendApplicationDefinedRTCPPacket(const unsigned char subType,
+Channel::SendApplicationDefinedRTCPPacket(unsigned char subType,
                                              unsigned int name,
                                              const char* data,
                                              unsigned short dataLengthInBytes)
@@ -4835,7 +4835,7 @@ Channel::GetRtpRtcp(RtpRtcp* &rtpRtcpModule) const
 // TODO(andrew): refactor Mix functions here and in transmit_mixer.cc to use
 // a shared helper.
 int32_t
-Channel::MixOrReplaceAudioWithFile(const int mixingFrequency)
+Channel::MixOrReplaceAudioWithFile(int mixingFrequency)
 {
     scoped_array<int16_t> fileBuffer(new int16_t[640]);
     int fileSamples(0);
@@ -4903,7 +4903,7 @@ Channel::MixOrReplaceAudioWithFile(const int mixingFrequency)
 
 int32_t
 Channel::MixAudioWithFile(AudioFrame& audioFrame,
-                          const int mixingFrequency)
+                          int mixingFrequency)
 {
     assert(mixingFrequency <= 32000);
 
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 582a6fbf4..4826ee6b8 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -72,9 +72,9 @@ public:
 public:
     virtual ~Channel();
     static int32_t CreateChannel(Channel*& channel,
-                                 const int32_t channelId,
-                                 const uint32_t instanceId);
-    Channel(const int32_t channelId, const uint32_t instanceId);
+                                 int32_t channelId,
+                                 uint32_t instanceId);
+    Channel(int32_t channelId, uint32_t instanceId);
     int32_t Init();
     int32_t SetEngineInformation(
         Statistics& engineStatistics,
@@ -139,37 +139,37 @@ public:
     int32_t GetPeriodicDeadOrAliveStatus(bool& enabled, int& sampleTimeSeconds);
 
     // VoEFile
-    int StartPlayingFileLocally(const char* fileName, const bool loop,
-                                const FileFormats format,
-                                const int startPosition,
-                                const float volumeScaling,
-                                const int stopPosition,
+    int StartPlayingFileLocally(const char* fileName, bool loop,
+                                FileFormats format,
+                                int startPosition,
+                                float volumeScaling,
+                                int stopPosition,
                                 const CodecInst* codecInst);
-    int StartPlayingFileLocally(InStream* stream, const FileFormats format,
-                                const int startPosition,
-                                const float volumeScaling,
-                                const int stopPosition,
+    int StartPlayingFileLocally(InStream* stream, FileFormats format,
+                                int startPosition,
+                                float volumeScaling,
+                                int stopPosition,
                                 const CodecInst* codecInst);
     int StopPlayingFileLocally();
     int IsPlayingFileLocally() const;
     int RegisterFilePlayingToMixer();
-    int ScaleLocalFilePlayout(const float scale);
+    int ScaleLocalFilePlayout(float scale);
     int GetLocalPlayoutPosition(int& positionMs);
-    int StartPlayingFileAsMicrophone(const char* fileName, const bool loop,
-                                     const FileFormats format,
-                                     const int startPosition,
-                                     const float volumeScaling,
-                                     const int stopPosition,
+    int StartPlayingFileAsMicrophone(const char* fileName, bool loop,
+                                     FileFormats format,
+                                     int startPosition,
+                                     float volumeScaling,
+                                     int stopPosition,
                                      const CodecInst* codecInst);
     int StartPlayingFileAsMicrophone(InStream* stream,
-                                     const FileFormats format,
-                                     const int startPosition,
-                                     const float volumeScaling,
-                                     const int stopPosition,
+                                     FileFormats format,
+                                     int startPosition,
+                                     float volumeScaling,
+                                     int stopPosition,
                                      const CodecInst* codecInst);
     int StopPlayingFileAsMicrophone();
     int IsPlayingFileAsMicrophone() const;
-    int ScaleFileAsMicrophonePlayout(const float scale);
+    int ScaleFileAsMicrophonePlayout(float scale);
     int StartRecordingPlayout(const char* fileName, const CodecInst* codecInst);
     int StartRecordingPlayout(OutStream* stream, const CodecInst* codecInst);
     int StopRecordingPlayout();
@@ -185,7 +185,7 @@ public:
     // VoEVolumeControl
     int GetSpeechOutputLevel(uint32_t& level) const;
     int GetSpeechOutputLevelFullRange(uint32_t& level) const;
-    int SetMute(const bool enable);
+    int SetMute(bool enable);
     bool Mute() const;
     int SetOutputVolumePan(float left, float right);
     int GetOutputVolumePan(float& left, float& right) const;
@@ -234,13 +234,13 @@ public:
     int DeRegisterRxVadObserver();
     int VoiceActivityIndicator(int &activity);
 #ifdef WEBRTC_VOICE_ENGINE_AGC
-    int SetRxAgcStatus(const bool enable, const AgcModes mode);
+    int SetRxAgcStatus(bool enable, AgcModes mode);
     int GetRxAgcStatus(bool& enabled, AgcModes& mode);
-    int SetRxAgcConfig(const AgcConfig config);
+    int SetRxAgcConfig(AgcConfig config);
     int GetRxAgcConfig(AgcConfig& config);
 #endif
 #ifdef WEBRTC_VOICE_ENGINE_NR
-    int SetRxNsStatus(const bool enable, const NsModes mode);
+    int SetRxNsStatus(bool enable, NsModes mode);
     int GetRxNsStatus(bool& enabled, NsModes& mode);
 #endif
 
@@ -264,7 +264,7 @@ public:
                           unsigned int& timestamp,
                           unsigned int& playoutTimestamp, unsigned int* jitter,
                           unsigned short* fractionLost);
-    int SendApplicationDefinedRTCPPacket(const unsigned char subType,
+    int SendApplicationDefinedRTCPPacket(unsigned char subType,
                                          unsigned int name, const char* data,
                                          unsigned short dataLengthInBytes);
     int GetRTPStatistics(unsigned int& averageJitterMs,
@@ -295,55 +295,55 @@ public:
     int32_t InFrameType(int16_t frameType);
 
 public:
-    int32_t OnRxVadDetected(const int vadDecision);
+    int32_t OnRxVadDetected(int vadDecision);
 
 public:
     // From RtpData in the RTP/RTCP module
     int32_t OnReceivedPayloadData(const uint8_t* payloadData,
-                                  const uint16_t payloadSize,
+                                  uint16_t payloadSize,
                                   const WebRtcRTPHeader* rtpHeader);
 
 public:
     // From RtpFeedback in the RTP/RTCP module
     int32_t OnInitializeDecoder(
-            const int32_t id,
-            const int8_t payloadType,
+            int32_t id,
+            int8_t payloadType,
             const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-            const int frequency,
-            const uint8_t channels,
-            const uint32_t rate);
+            int frequency,
+            uint8_t channels,
+            uint32_t rate);
 
-    void OnPacketTimeout(const int32_t id);
+    void OnPacketTimeout(int32_t id);
 
-    void OnReceivedPacket(const int32_t id, const RtpRtcpPacketType packetType);
+    void OnReceivedPacket(int32_t id, RtpRtcpPacketType packetType);
 
-    void OnPeriodicDeadOrAlive(const int32_t id,
-                               const RTPAliveType alive);
+    void OnPeriodicDeadOrAlive(int32_t id,
+                               RTPAliveType alive);
 
-    void OnIncomingSSRCChanged(const int32_t id,
-                               const uint32_t SSRC);
+    void OnIncomingSSRCChanged(int32_t id,
+                               uint32_t SSRC);
 
-    void OnIncomingCSRCChanged(const int32_t id,
-                               const uint32_t CSRC, const bool added);
+    void OnIncomingCSRCChanged(int32_t id,
+                               uint32_t CSRC, bool added);
 
 public:
     // From RtcpFeedback in the RTP/RTCP module
-    void OnApplicationDataReceived(const int32_t id,
-                                   const uint8_t subType,
-                                   const uint32_t name,
-                                   const uint16_t length,
+    void OnApplicationDataReceived(int32_t id,
+                                   uint8_t subType,
+                                   uint32_t name,
+                                   uint16_t length,
                                    const uint8_t* data);
 
 public:
     // From RtpAudioFeedback in the RTP/RTCP module
-    void OnReceivedTelephoneEvent(const int32_t id,
-                                  const uint8_t event,
-                                  const bool endOfEvent);
+    void OnReceivedTelephoneEvent(int32_t id,
+                                  uint8_t event,
+                                  bool endOfEvent);
 
-    void OnPlayTelephoneEvent(const int32_t id,
-                              const uint8_t event,
-                              const uint16_t lengthMs,
-                              const uint8_t volume);
+    void OnPlayTelephoneEvent(int32_t id,
+                              uint8_t event,
+                              uint16_t lengthMs,
+                              uint8_t volume);
 
 public:
     // From Transport (called by the RTP/RTCP module)
@@ -352,8 +352,8 @@ public:
 
 public:
     // From MixerParticipant
-    int32_t GetAudioFrame(const int32_t id, AudioFrame& audioFrame);
-    int32_t NeededFrequency(const int32_t id);
+    int32_t GetAudioFrame(int32_t id, AudioFrame& audioFrame);
+    int32_t NeededFrequency(int32_t id);
 
 public:
     // From MonitorObserver
@@ -361,12 +361,12 @@ public:
 
 public:
     // From FileCallback
-    void PlayNotification(const int32_t id,
-                          const uint32_t durationMs);
-    void RecordNotification(const int32_t id,
-                            const uint32_t durationMs);
-    void PlayFileEnded(const int32_t id);
-    void RecordFileEnded(const int32_t id);
+    void PlayNotification(int32_t id,
+                          uint32_t durationMs);
+    void RecordNotification(int32_t id,
+                            uint32_t durationMs);
+    void PlayFileEnded(int32_t id);
+    void RecordFileEnded(int32_t id);
 
 public:
     uint32_t InstanceId() const
@@ -423,8 +423,8 @@ public:
 
 private:
     int InsertInbandDtmfTone();
-    int32_t MixOrReplaceAudioWithFile(const int mixingFrequency);
-    int32_t MixAudioWithFile(AudioFrame& audioFrame, const int mixingFrequency);
+    int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
+    int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
     void UpdateDeadOrAliveCounters(bool alive);
     int32_t SendPacketRaw(const void *data, int len, bool RTCP);
     void UpdatePacketDelay(uint32_t timestamp,
diff --git a/webrtc/voice_engine/channel_manager.cc b/webrtc/voice_engine/channel_manager.cc
index 72426e45f..770bde353 100644
--- a/webrtc/voice_engine/channel_manager.cc
+++ b/webrtc/voice_engine/channel_manager.cc
@@ -17,7 +17,7 @@ namespace webrtc
 namespace voe
 {
 
-ChannelManager::ChannelManager(const uint32_t instanceId) :
+ChannelManager::ChannelManager(uint32_t instanceId) :
     ChannelManagerBase(),
     _instanceId(instanceId)
 {
@@ -33,7 +33,7 @@ bool ChannelManager::CreateChannel(int32_t& channelId)
     return ChannelManagerBase::CreateItem(channelId);
 }
 
-int32_t ChannelManager::DestroyChannel(const int32_t channelId)
+int32_t ChannelManager::DestroyChannel(int32_t channelId)
 {
     Channel* deleteChannel =
         static_cast<Channel*> (ChannelManagerBase::RemoveItem(channelId));
@@ -71,7 +71,7 @@ void ChannelManager::DeleteItem(void* item)
     delete deleteItem;
 }
 
-Channel* ChannelManager::GetChannel(const int32_t channelId) const
+Channel* ChannelManager::GetChannel(int32_t channelId) const
 {
     return static_cast<Channel*> (ChannelManagerBase::GetItem(channelId));
 }
diff --git a/webrtc/voice_engine/channel_manager.h b/webrtc/voice_engine/channel_manager.h
index defade1c7..91247d6a2 100644
--- a/webrtc/voice_engine/channel_manager.h
+++ b/webrtc/voice_engine/channel_manager.h
@@ -30,7 +30,7 @@ class ChannelManager: private ChannelManagerBase
 public:
     bool CreateChannel(int32_t& channelId);
 
-    int32_t DestroyChannel(const int32_t channelId);
+    int32_t DestroyChannel(int32_t channelId);
 
     int32_t MaxNumOfChannels() const;
 
@@ -39,7 +39,7 @@ public:
     void GetChannelIds(int32_t* channelsArray,
                        int32_t& numOfChannels) const;
 
-    ChannelManager(const uint32_t instanceId);
+    ChannelManager(uint32_t instanceId);
 
     ~ChannelManager();
 
@@ -48,7 +48,7 @@ private:
 
     ChannelManager& operator=(const ChannelManager&);
 
-    Channel* GetChannel(const int32_t channelId) const;
+    Channel* GetChannel(int32_t channelId) const;
 
     void GetChannels(MapWrapper& channels) const;
 
diff --git a/webrtc/voice_engine/dtmf_inband.cc b/webrtc/voice_engine/dtmf_inband.cc
index c68970957..7a66aa064 100644
--- a/webrtc/voice_engine/dtmf_inband.cc
+++ b/webrtc/voice_engine/dtmf_inband.cc
@@ -64,7 +64,7 @@ const int16_t Dtmf_dBm0kHz[37]=
 };
 
 
-DtmfInband::DtmfInband(const int32_t id) :
+DtmfInband::DtmfInband(int32_t id) :
     _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _id(id),
     _outputFrequencyHz(8000),
@@ -87,7 +87,7 @@ DtmfInband::~DtmfInband()
 }
 
 int
-DtmfInband::SetSampleRate(const uint16_t frequency)
+DtmfInband::SetSampleRate(uint16_t frequency)
 {
     if (frequency != 8000 &&
             frequency != 16000 &&
@@ -125,7 +125,7 @@ DtmfInband::Init()
 }
 
 int
-DtmfInband::AddTone(const uint8_t eventCode,
+DtmfInband::AddTone(uint8_t eventCode,
                     int32_t lengthMs,
                     int32_t attenuationDb)
 {
@@ -170,7 +170,7 @@ DtmfInband::ResetTone()
 }
 
 int
-DtmfInband::StartTone(const uint8_t eventCode,
+DtmfInband::StartTone(uint8_t eventCode,
                       int32_t attenuationDb)
 {
     CriticalSectionScoped lock(&_critSect);
@@ -263,10 +263,10 @@ DtmfInband::DelaySinceLastTone() const
 
 int16_t
 DtmfInband::DtmfFix_generate(int16_t *decoded,
-                             const int16_t value,
-                             const int16_t volume,
-                             const int16_t frameLen,
-                             const int16_t fs)
+                             int16_t value,
+                             int16_t volume,
+                             int16_t frameLen,
+                             int16_t fs)
 {
     const int16_t *a_times2Tbl;
     const int16_t *y2_Table;
@@ -348,11 +348,11 @@ DtmfInband::DtmfFix_generate(int16_t *decoded,
 }
 
 int16_t
-DtmfInband::DtmfFix_generateSignal(const int16_t a1_times2,
-                                   const int16_t a2_times2,
-                                   const int16_t volume,
+DtmfInband::DtmfFix_generateSignal(int16_t a1_times2,
+                                   int16_t a2_times2,
+                                   int16_t volume,
                                    int16_t *signal,
-                                   const int16_t length)
+                                   int16_t length)
 {
     int i;
 
diff --git a/webrtc/voice_engine/dtmf_inband.h b/webrtc/voice_engine/dtmf_inband.h
index c0a431c4d..f1e523ff2 100644
--- a/webrtc/voice_engine/dtmf_inband.h
+++ b/webrtc/voice_engine/dtmf_inband.h
@@ -24,22 +24,22 @@ class CriticalSectionWrapper;
 class DtmfInband
 {
 public:
-    DtmfInband(const int32_t id);
+    DtmfInband(int32_t id);
 
     virtual ~DtmfInband();
 
     void Init();
 
-    int SetSampleRate(const uint16_t frequency);
+    int SetSampleRate(uint16_t frequency);
 
     int GetSampleRate(uint16_t& frequency);
 
-    int AddTone(const uint8_t eventCode,
+    int AddTone(uint8_t eventCode,
                 int32_t lengthMs,
                 int32_t attenuationDb);
 
     int ResetTone();
-    int StartTone(const uint8_t eventCode, int32_t attenuationDb);
+    int StartTone(uint8_t eventCode, int32_t attenuationDb);
 
     int StopTone();
 
@@ -54,21 +54,21 @@ public:
 private:
     void ReInit();
     int16_t DtmfFix_generate(int16_t* decoded,
-                             const int16_t value,
-                             const int16_t volume,
-                             const int16_t frameLen,
-                             const int16_t fs);
+                             int16_t value,
+                             int16_t volume,
+                             int16_t frameLen,
+                             int16_t fs);
 
 private:
     enum {kDtmfFrameSizeMs = 10};
     enum {kDtmfAmpHigh = 32768};
     enum {kDtmfAmpLow  = 23171};	// 3 dB lower than the high frequency
 
-    int16_t DtmfFix_generateSignal(const int16_t a1_times2,
-                                   const int16_t a2_times2,
-                                   const int16_t volume,
+    int16_t DtmfFix_generateSignal(int16_t a1_times2,
+                                   int16_t a2_times2,
+                                   int16_t volume,
                                    int16_t* signal,
-                                   const int16_t length);
+                                   int16_t length);
 
 private:
     CriticalSectionWrapper& _critSect;
diff --git a/webrtc/voice_engine/dtmf_inband_queue.cc b/webrtc/voice_engine/dtmf_inband_queue.cc
index fa1dce134..4e11a2470 100644
--- a/webrtc/voice_engine/dtmf_inband_queue.cc
+++ b/webrtc/voice_engine/dtmf_inband_queue.cc
@@ -13,7 +13,7 @@
 
 namespace webrtc {
 
-DtmfInbandQueue::DtmfInbandQueue(const int32_t id):
+DtmfInbandQueue::DtmfInbandQueue(int32_t id):
     _id(id),
     _DtmfCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
     _nextEmptyIndex(0)
diff --git a/webrtc/voice_engine/dtmf_inband_queue.h b/webrtc/voice_engine/dtmf_inband_queue.h
index 6a65c9e0a..98e4bee7d 100644
--- a/webrtc/voice_engine/dtmf_inband_queue.h
+++ b/webrtc/voice_engine/dtmf_inband_queue.h
@@ -22,7 +22,7 @@ class DtmfInbandQueue
 {
 public:
 
-    DtmfInbandQueue(const int32_t id);
+    DtmfInbandQueue(int32_t id);
 
     virtual ~DtmfInbandQueue();
 
diff --git a/webrtc/voice_engine/include/mock/mock_voe_connection_observer.h b/webrtc/voice_engine/include/mock/mock_voe_connection_observer.h
index 62e572e76..12ca63956 100644
--- a/webrtc/voice_engine/include/mock/mock_voe_connection_observer.h
+++ b/webrtc/voice_engine/include/mock/mock_voe_connection_observer.h
@@ -17,8 +17,8 @@ namespace webrtc {
 
 class MockVoeConnectionObserver : public VoEConnectionObserver {
  public:
-  MOCK_METHOD2(OnPeriodicDeadOrAlive, void(const int channel,
-                                           const bool alive));
+  MOCK_METHOD2(OnPeriodicDeadOrAlive, void(int channel,
+                                           bool alive));
 };
 
 }
diff --git a/webrtc/voice_engine/include/mock/mock_voe_observer.h b/webrtc/voice_engine/include/mock/mock_voe_observer.h
index c01320d6b..839a9b568 100644
--- a/webrtc/voice_engine/include/mock/mock_voe_observer.h
+++ b/webrtc/voice_engine/include/mock/mock_voe_observer.h
@@ -21,7 +21,7 @@ class MockVoEObserver: public VoiceEngineObserver {
   MockVoEObserver() {}
   virtual ~MockVoEObserver() {}
 
-  MOCK_METHOD2(CallbackOnError, void(const int channel, const int error_code));
+  MOCK_METHOD2(CallbackOnError, void(int channel, int error_code));
 };
 
 }
diff --git a/webrtc/voice_engine/include/voe_audio_processing.h b/webrtc/voice_engine/include/voe_audio_processing.h
index 4965d331b..cc62eb904 100644
--- a/webrtc/voice_engine/include/voe_audio_processing.h
+++ b/webrtc/voice_engine/include/voe_audio_processing.h
@@ -83,7 +83,7 @@ public:
     // Sets the AGC configuration.
     // Should only be used in situations where the working environment
     // is well known.
-    virtual int SetAgcConfig(const AgcConfig config) = 0;
+    virtual int SetAgcConfig(AgcConfig config) = 0;
 
     // Gets the AGC configuration.
     virtual int GetAgcConfig(AgcConfig& config) = 0;
@@ -152,7 +152,7 @@ public:
 
     // Modifies the AGC configuration on the receiving side for the
     // specified |channel|.
-    virtual int SetRxAgcConfig(int channel, const AgcConfig config) = 0;
+    virtual int SetRxAgcConfig(int channel, AgcConfig config) = 0;
 
     // Gets the AGC configuration on the receiving side.
     virtual int GetRxAgcConfig(int channel, AgcConfig& config) = 0;
diff --git a/webrtc/voice_engine/include/voe_base.h b/webrtc/voice_engine/include/voe_base.h
index 70cbab191..529ea6b6d 100644
--- a/webrtc/voice_engine/include/voe_base.h
+++ b/webrtc/voice_engine/include/voe_base.h
@@ -50,7 +50,7 @@ public:
     // This method will be called after the occurrence of any runtime error
     // code, or warning notification, when the observer interface has been
     // installed using VoEBase::RegisterVoiceEngineObserver().
-    virtual void CallbackOnError(const int channel, const int errCode) = 0;
+    virtual void CallbackOnError(int channel, int errCode) = 0;
 
 protected:
     virtual ~VoiceEngineObserver() {}
@@ -72,11 +72,11 @@ public:
 
     // Specifies the amount and type of trace information which will be
     // created by the VoiceEngine.
-    static int SetTraceFilter(const unsigned int filter);
+    static int SetTraceFilter(unsigned int filter);
 
     // Sets the name of the trace file and enables non-encrypted trace messages.
     static int SetTraceFile(const char* fileNameUTF8,
-                            const bool addFileCounter = false);
+                            bool addFileCounter = false);
 
     // Installs the TraceCallback implementation to ensure that the user
     // receives callbacks for generated trace messages.
diff --git a/webrtc/voice_engine/include/voe_external_media.h b/webrtc/voice_engine/include/voe_external_media.h
index d4aa8f078..a3d1311dd 100644
--- a/webrtc/voice_engine/include/voe_external_media.h
+++ b/webrtc/voice_engine/include/voe_external_media.h
@@ -51,9 +51,9 @@ public:
     // The sampling frequency will depend upon the codec used.
     // If |isStereo| is true, audio10ms will contain 16-bit PCM data
     // samples in interleaved stereo format (L0,R0,L1,R1,...).
-    virtual void Process(const int channel, const ProcessingTypes type,
-                         int16_t audio10ms[], const int length,
-                         const int samplingFreq, const bool isStereo) = 0;
+    virtual void Process(int channel, ProcessingTypes type,
+                         int16_t audio10ms[], int length,
+                         int samplingFreq, bool isStereo) = 0;
 
 protected:
     virtual ~VoEMediaProcess() {}
diff --git a/webrtc/voice_engine/include/voe_network.h b/webrtc/voice_engine/include/voe_network.h
index 1e0874a1a..6ab92264b 100644
--- a/webrtc/voice_engine/include/voe_network.h
+++ b/webrtc/voice_engine/include/voe_network.h
@@ -47,7 +47,7 @@ public:
     // This method will be called peridically and deliver dead-or-alive
     // notifications for a specified |channel| when the observer interface
     // has been installed and activated.
-    virtual void OnPeriodicDeadOrAlive(const int channel, const bool alive) = 0;
+    virtual void OnPeriodicDeadOrAlive(int channel, bool alive) = 0;
 
 protected:
     virtual ~VoEConnectionObserver() {}
diff --git a/webrtc/voice_engine/include/voe_rtp_rtcp.h b/webrtc/voice_engine/include/voe_rtp_rtcp.h
index 4f22c59a1..59ca61af7 100644
--- a/webrtc/voice_engine/include/voe_rtp_rtcp.h
+++ b/webrtc/voice_engine/include/voe_rtp_rtcp.h
@@ -52,10 +52,10 @@ class WEBRTC_DLLEXPORT VoERTPObserver
 {
 public:
     virtual void OnIncomingCSRCChanged(
-        const int channel, const unsigned int CSRC, const bool added) = 0;
+        int channel, unsigned int CSRC, bool added) = 0;
 
     virtual void OnIncomingSSRCChanged(
-        const int channel, const unsigned int SSRC) = 0;
+        int channel, unsigned int SSRC) = 0;
 
 protected:
     virtual ~VoERTPObserver() {}
@@ -66,9 +66,9 @@ class WEBRTC_DLLEXPORT VoERTCPObserver
 {
 public:
     virtual void OnApplicationDataReceived(
-        const int channel, const unsigned char subType,
-        const unsigned int name, const unsigned char* data,
-        const unsigned short dataLengthInBytes) = 0;
+        int channel, unsigned char subType,
+        unsigned int name, const unsigned char* data,
+        unsigned short dataLengthInBytes) = 0;
 
 protected:
     virtual ~VoERTCPObserver() {}
@@ -209,7 +209,7 @@ public:
 
     // Sends an RTCP APP packet on a specific |channel|.
     virtual int SendApplicationDefinedRTCPPacket(
-        int channel, const unsigned char subType, unsigned int name,
+        int channel, unsigned char subType, unsigned int name,
         const char* data, unsigned short dataLengthInBytes) = 0;
 
     // Sets the Forward Error Correction (FEC) status on a specific |channel|.
diff --git a/webrtc/voice_engine/monitor_module.cc b/webrtc/voice_engine/monitor_module.cc
index 08fe58017..ef863f90b 100644
--- a/webrtc/voice_engine/monitor_module.cc
+++ b/webrtc/voice_engine/monitor_module.cc
@@ -61,7 +61,7 @@ MonitorModule::Version(char* version,
 }
    
 int32_t 
-MonitorModule::ChangeUniqueId(const int32_t id)
+MonitorModule::ChangeUniqueId(int32_t id)
 {
     return 0;
 }
diff --git a/webrtc/voice_engine/monitor_module.h b/webrtc/voice_engine/monitor_module.h
index e381b7c1f..c82b4ee7d 100644
--- a/webrtc/voice_engine/monitor_module.h
+++ b/webrtc/voice_engine/monitor_module.h
@@ -44,7 +44,7 @@ public:	// module
                     uint32_t& remainingBufferInBytes,
                     uint32_t& position) const;
 
-    int32_t ChangeUniqueId(const int32_t id);
+    int32_t ChangeUniqueId(int32_t id);
 
     int32_t TimeUntilNextProcess();
 
diff --git a/webrtc/voice_engine/output_mixer.cc b/webrtc/voice_engine/output_mixer.cc
index a8e417793..b928455b1 100644
--- a/webrtc/voice_engine/output_mixer.cc
+++ b/webrtc/voice_engine/output_mixer.cc
@@ -24,10 +24,10 @@ namespace webrtc {
 namespace voe {
 
 void
-OutputMixer::NewMixedAudio(const int32_t id,
+OutputMixer::NewMixedAudio(int32_t id,
                            const AudioFrame& generalAudioFrame,
                            const AudioFrame** uniqueAudioFrames,
-                           const uint32_t size)
+                           uint32_t size)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
                  "OutputMixer::NewMixedAudio(id=%d, size=%u)", id, size);
@@ -37,29 +37,29 @@ OutputMixer::NewMixedAudio(const int32_t id,
 }
 
 void OutputMixer::MixedParticipants(
-    const int32_t id,
+    int32_t id,
     const ParticipantStatistics* participantStatistics,
-    const uint32_t size)
+    uint32_t size)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
                  "OutputMixer::MixedParticipants(id=%d, size=%u)", id, size);
 }
 
-void OutputMixer::VADPositiveParticipants(const int32_t id,
-    const ParticipantStatistics* participantStatistics, const uint32_t size)
+void OutputMixer::VADPositiveParticipants(int32_t id,
+    const ParticipantStatistics* participantStatistics, uint32_t size)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
                  "OutputMixer::VADPositiveParticipants(id=%d, size=%u)",
                  id, size);
 }
 
-void OutputMixer::MixedAudioLevel(const int32_t id, const uint32_t level)
+void OutputMixer::MixedAudioLevel(int32_t id, uint32_t level)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
                  "OutputMixer::MixedAudioLevel(id=%d, level=%u)", id, level);
 }
 
-void OutputMixer::PlayNotification(const int32_t id, const uint32_t durationMs)
+void OutputMixer::PlayNotification(int32_t id, uint32_t durationMs)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
                  "OutputMixer::PlayNotification(id=%d, durationMs=%d)",
@@ -67,8 +67,8 @@ void OutputMixer::PlayNotification(const int32_t id, const uint32_t durationMs)
     // Not implement yet
 }
 
-void OutputMixer::RecordNotification(const int32_t id,
-                                     const uint32_t durationMs)
+void OutputMixer::RecordNotification(int32_t id,
+                                     uint32_t durationMs)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
                  "OutputMixer::RecordNotification(id=%d, durationMs=%d)",
@@ -77,7 +77,7 @@ void OutputMixer::RecordNotification(const int32_t id,
     // Not implement yet
 }
 
-void OutputMixer::PlayFileEnded(const int32_t id)
+void OutputMixer::PlayFileEnded(int32_t id)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
                  "OutputMixer::PlayFileEnded(id=%d)", id);
@@ -85,7 +85,7 @@ void OutputMixer::PlayFileEnded(const int32_t id)
     // not needed
 }
 
-void OutputMixer::RecordFileEnded(const int32_t id)
+void OutputMixer::RecordFileEnded(int32_t id)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
                  "OutputMixer::RecordFileEnded(id=%d)", id);
@@ -99,7 +99,7 @@ void OutputMixer::RecordFileEnded(const int32_t id)
 }
 
 int32_t
-OutputMixer::Create(OutputMixer*& mixer, const uint32_t instanceId)
+OutputMixer::Create(OutputMixer*& mixer, uint32_t instanceId)
 {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId,
                  "OutputMixer::Create(instanceId=%d)", instanceId);
@@ -114,7 +114,7 @@ OutputMixer::Create(OutputMixer*& mixer, const uint32_t instanceId)
     return 0;
 }
 
-OutputMixer::OutputMixer(const uint32_t instanceId) :
+OutputMixer::OutputMixer(uint32_t instanceId) :
     _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _mixerModule(*AudioConferenceMixer::Create(instanceId)),
@@ -262,14 +262,14 @@ int OutputMixer::StopPlayingDtmfTone()
 
 int32_t
 OutputMixer::SetMixabilityStatus(MixerParticipant& participant,
-                                 const bool mixable)
+                                 bool mixable)
 {
     return _mixerModule.SetMixabilityStatus(participant, mixable);
 }
 
 int32_t
 OutputMixer::SetAnonymousMixabilityStatus(MixerParticipant& participant,
-                                          const bool mixable)
+                                          bool mixable)
 {
     return _mixerModule.SetAnonymousMixabilityStatus(participant,mixable);
 }
diff --git a/webrtc/voice_engine/output_mixer.h b/webrtc/voice_engine/output_mixer.h
index b98f88ead..5a423c2ac 100644
--- a/webrtc/voice_engine/output_mixer.h
+++ b/webrtc/voice_engine/output_mixer.h
@@ -36,7 +36,7 @@ class OutputMixer : public AudioMixerOutputReceiver,
                     public FileCallback
 {
 public:
-    static int32_t Create(OutputMixer*& mixer, const uint32_t instanceId);
+    static int32_t Create(OutputMixer*& mixer, uint32_t instanceId);
 
     static void Destroy(OutputMixer*& mixer);
 
@@ -63,10 +63,10 @@ public:
     int32_t DoOperationsOnCombinedSignal();
 
     int32_t SetMixabilityStatus(MixerParticipant& participant,
-                                const bool mixable);
+                                bool mixable);
 
     int32_t SetAnonymousMixabilityStatus(MixerParticipant& participant,
-                                         const bool mixable);
+                                         bool mixable);
 
     int GetMixedAudio(int sample_rate_hz, int num_channels,
                       AudioFrame* audioFrame);
@@ -92,34 +92,34 @@ public:
 
     // from AudioMixerOutputReceiver
     virtual void NewMixedAudio(
-        const int32_t id,
+        int32_t id,
         const AudioFrame& generalAudioFrame,
         const AudioFrame** uniqueAudioFrames,
-        const uint32_t size);
+        uint32_t size);
 
     // from AudioMixerStatusReceiver
     virtual void MixedParticipants(
-        const int32_t id,
+        int32_t id,
         const ParticipantStatistics* participantStatistics,
-        const uint32_t size);
+        uint32_t size);
 
     virtual void VADPositiveParticipants(
-        const int32_t id,
+        int32_t id,
         const ParticipantStatistics* participantStatistics,
-        const uint32_t size);
+        uint32_t size);
 
-    virtual void MixedAudioLevel(const int32_t id, const uint32_t level);
+    virtual void MixedAudioLevel(int32_t id, uint32_t level);
 
     // For file recording
-    void PlayNotification(const int32_t id, const uint32_t durationMs);
+    void PlayNotification(int32_t id, uint32_t durationMs);
 
-    void RecordNotification(const int32_t id, const uint32_t durationMs);
+    void RecordNotification(int32_t id, uint32_t durationMs);
 
-    void PlayFileEnded(const int32_t id);
-    void RecordFileEnded(const int32_t id);
+    void PlayFileEnded(int32_t id);
+    void RecordFileEnded(int32_t id);
 
 private:
-    OutputMixer(const uint32_t instanceId);
+    OutputMixer(uint32_t instanceId);
     void APMAnalyzeReverseStream();
     int InsertInbandDtmfTone();
 
diff --git a/webrtc/voice_engine/shared_data.cc b/webrtc/voice_engine/shared_data.cc
index 1393eb33a..2f0b04a45 100644
--- a/webrtc/voice_engine/shared_data.cc
+++ b/webrtc/voice_engine/shared_data.cc
@@ -104,16 +104,16 @@ uint16_t SharedData::NumOfSendingChannels()
     return nChannelsSending;
 }
 
-void SharedData::SetLastError(const int32_t error) const {
+void SharedData::SetLastError(int32_t error) const {
   _engineStatistics.SetLastError(error);
 }
 
-void SharedData::SetLastError(const int32_t error,
-                              const TraceLevel level) const {
+void SharedData::SetLastError(int32_t error,
+                              TraceLevel level) const {
   _engineStatistics.SetLastError(error, level);
 }
 
-void SharedData::SetLastError(const int32_t error, const TraceLevel level,
+void SharedData::SetLastError(int32_t error, TraceLevel level,
                               const char* msg) const {
   _engineStatistics.SetLastError(error, level, msg);
 }
diff --git a/webrtc/voice_engine/shared_data.h b/webrtc/voice_engine/shared_data.h
index 99cff04cc..b20225f08 100644
--- a/webrtc/voice_engine/shared_data.h
+++ b/webrtc/voice_engine/shared_data.h
@@ -58,9 +58,9 @@ public:
     uint16_t NumOfSendingChannels();
 
     // Convenience methods for calling statistics().SetLastError().
-    void SetLastError(const int32_t error) const;
-    void SetLastError(const int32_t error, const TraceLevel level) const;
-    void SetLastError(const int32_t error, const TraceLevel level,
+    void SetLastError(int32_t error) const;
+    void SetLastError(int32_t error, TraceLevel level) const;
+    void SetLastError(int32_t error, TraceLevel level,
                       const char* msg) const;
 
 protected:
diff --git a/webrtc/voice_engine/statistics.cc b/webrtc/voice_engine/statistics.cc
index 8003d6dbf..49861e0d3 100644
--- a/webrtc/voice_engine/statistics.cc
+++ b/webrtc/voice_engine/statistics.cc
@@ -20,7 +20,7 @@ namespace webrtc {
 
 namespace voe {
 
-Statistics::Statistics(const uint32_t instanceId) :
+Statistics::Statistics(uint32_t instanceId) :
     _critPtr(CriticalSectionWrapper::CreateCriticalSection()),
     _instanceId(instanceId),
     _lastError(0),
@@ -54,15 +54,15 @@ bool Statistics::Initialized() const
     return _isInitialized;
 }
 
-int32_t Statistics::SetLastError(const int32_t error) const
+int32_t Statistics::SetLastError(int32_t error) const
 {
     CriticalSectionScoped cs(_critPtr);
     _lastError = error;
     return 0;
 }
 
-int32_t Statistics::SetLastError(const int32_t error,
-                                 const TraceLevel level) const
+int32_t Statistics::SetLastError(int32_t error,
+                                 TraceLevel level) const
 {
     CriticalSectionScoped cs(_critPtr);
     _lastError = error;
@@ -73,8 +73,8 @@ int32_t Statistics::SetLastError(const int32_t error,
 }
 
 int32_t Statistics::SetLastError(
-    const int32_t error,
-    const TraceLevel level, const char* msg) const
+    int32_t error,
+    TraceLevel level, const char* msg) const
 {
     CriticalSectionScoped cs(_critPtr);
     char traceMessage[KTraceMaxMessageSize];
diff --git a/webrtc/voice_engine/statistics.h b/webrtc/voice_engine/statistics.h
index 057f26629..369a79a88 100644
--- a/webrtc/voice_engine/statistics.h
+++ b/webrtc/voice_engine/statistics.h
@@ -26,16 +26,16 @@ class Statistics
  public:
     enum {KTraceMaxMessageSize = 256};
  public:
-    Statistics(const uint32_t instanceId);
+    Statistics(uint32_t instanceId);
     ~Statistics();
 
     int32_t SetInitialized();
     int32_t SetUnInitialized();
     bool Initialized() const;
-    int32_t SetLastError(const int32_t error) const;
-    int32_t SetLastError(const int32_t error, const TraceLevel level) const;
-    int32_t SetLastError(const int32_t error,
-                         const TraceLevel level,
+    int32_t SetLastError(int32_t error) const;
+    int32_t SetLastError(int32_t error, TraceLevel level) const;
+    int32_t SetLastError(int32_t error,
+                         TraceLevel level,
                          const char* msg) const;
     int32_t LastError() const;
 
diff --git a/webrtc/voice_engine/test/auto_test/fakes/fake_media_process.h b/webrtc/voice_engine/test/auto_test/fakes/fake_media_process.h
index edd80fc79..f3d57db95 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/fake_media_process.h
+++ b/webrtc/voice_engine/test/auto_test/fakes/fake_media_process.h
@@ -14,12 +14,12 @@
 
 class FakeMediaProcess : public webrtc::VoEMediaProcess {
  public:
-  virtual void Process(const int channel,
+  virtual void Process(int channel,
                        const webrtc::ProcessingTypes type,
                        int16_t audio_10ms[],
-                       const int length,
-                       const int sampling_freq_hz,
-                       const bool stereo) {
+                       int length,
+                       int sampling_freq_hz,
+                       bool stereo) {
     for (int i = 0; i < length; i++) {
       if (!stereo) {
         audio_10ms[i] = static_cast<int16_t>(audio_10ms[i] *
diff --git a/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.cc b/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.cc
index f0e665bc6..52c180b23 100644
--- a/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.cc
+++ b/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.cc
@@ -14,7 +14,7 @@ class TestErrorObserver : public webrtc::VoiceEngineObserver {
  public:
   TestErrorObserver() {}
   virtual ~TestErrorObserver() {}
-  void CallbackOnError(const int channel, const int error_code) {
+  void CallbackOnError(int channel, int error_code) {
     ADD_FAILURE() << "Unexpected error on channel " << channel <<
         ": error code " << error_code;
   }
diff --git a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
index c0f1b95c7..540c24b78 100644
--- a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
@@ -16,11 +16,11 @@ class TestRtpObserver : public webrtc::VoERTPObserver {
  public:
   TestRtpObserver();
   virtual ~TestRtpObserver();
-  virtual void OnIncomingCSRCChanged(const int channel,
-                                     const unsigned int CSRC,
-                                     const bool added);
-  virtual void OnIncomingSSRCChanged(const int channel,
-                                     const unsigned int SSRC);
+  virtual void OnIncomingCSRCChanged(int channel,
+                                     unsigned int CSRC,
+                                     bool added);
+  virtual void OnIncomingSSRCChanged(int channel,
+                                     unsigned int SSRC);
   void Reset();
  public:
   unsigned int ssrc_[2];
@@ -47,9 +47,9 @@ void TestRtpObserver::Reset() {
   }
 }
 
-void TestRtpObserver::OnIncomingCSRCChanged(const int channel,
-                                            const unsigned int CSRC,
-                                            const bool added) {
+void TestRtpObserver::OnIncomingCSRCChanged(int channel,
+                                            unsigned int CSRC,
+                                            bool added) {
   char msg[128];
   sprintf(msg, "=> OnIncomingCSRCChanged(channel=%d, CSRC=%u, added=%d)\n",
           channel, CSRC, added);
@@ -66,8 +66,8 @@ void TestRtpObserver::OnIncomingCSRCChanged(const int channel,
     size_[channel] = 0;
 }
 
-void TestRtpObserver::OnIncomingSSRCChanged(const int channel,
-                                            const unsigned int SSRC) {
+void TestRtpObserver::OnIncomingSSRCChanged(int channel,
+                                            unsigned int SSRC) {
   char msg[128];
   sprintf(msg, "\n=> OnIncomingSSRCChanged(channel=%d, SSRC=%u)\n", channel,
           SSRC);
@@ -78,11 +78,11 @@ void TestRtpObserver::OnIncomingSSRCChanged(const int channel,
 
 class RtcpAppHandler : public webrtc::VoERTCPObserver {
  public:
-  void OnApplicationDataReceived(const int channel,
-                                 const unsigned char sub_type,
-                                 const unsigned int name,
+  void OnApplicationDataReceived(int channel,
+                                 unsigned char sub_type,
+                                 unsigned int name,
                                  const unsigned char* data,
-                                 const unsigned short length_in_bytes);
+                                 unsigned short length_in_bytes);
   void Reset();
   ~RtcpAppHandler() {}
   unsigned short length_in_bytes_;
@@ -125,9 +125,9 @@ class RtpRtcpTest : public AfterStreamingFixture {
 };
 
 void RtcpAppHandler::OnApplicationDataReceived(
-    const int /*channel*/, const unsigned char sub_type,
-    const unsigned int name, const unsigned char* data,
-    const unsigned short length_in_bytes) {
+    const int /*channel*/, unsigned char sub_type,
+    unsigned int name, const unsigned char* data,
+    unsigned short length_in_bytes) {
   length_in_bytes_ = length_in_bytes;
   memcpy(data_, &data[0], length_in_bytes);
   sub_type_ = sub_type;
diff --git a/webrtc/voice_engine/test/auto_test/voe_extended_test.cc b/webrtc/voice_engine/test/auto_test/voe_extended_test.cc
index 209ff205b..90d12d6b3 100644
--- a/webrtc/voice_engine/test/auto_test/voe_extended_test.cc
+++ b/webrtc/voice_engine/test/auto_test/voe_extended_test.cc
@@ -169,7 +169,7 @@ void XRTPObserver::OnIncomingCSRCChanged(const int /*channel*/, const unsigned i
                                          const bool /*added*/) {
 }
 
-void XRTPObserver::OnIncomingSSRCChanged(const int /*channel*/, const unsigned int SSRC) {
+void XRTPObserver::OnIncomingSSRCChanged(const int /*channel*/, unsigned int SSRC) {
   // char msg[128];
   // sprintf(msg, "OnIncomingSSRCChanged(channel=%d, SSRC=%lu)\n",
   //        channel, SSRC);
@@ -199,7 +199,7 @@ int VoEExtendedTest::TestPassed(const char* str) const {
   return 0;
 }
 
-void VoEExtendedTest::OnPeriodicDeadOrAlive(const int /*channel*/, const bool alive) {
+void VoEExtendedTest::OnPeriodicDeadOrAlive(const int /*channel*/, bool alive) {
   _alive = alive;
   if (alive) {
     TEST_LOG("=> ALIVE ");
@@ -209,7 +209,7 @@ void VoEExtendedTest::OnPeriodicDeadOrAlive(const int /*channel*/, const bool al
   fflush(NULL);
 }
 
-void VoEExtendedTest::CallbackOnError(const int errCode, int) {
+void VoEExtendedTest::CallbackOnError(int errCode, int) {
   _errCode = errCode;
   TEST_LOG("\n************************\n");
   TEST_LOG(" RUNTIME ERROR: %d \n", errCode);
diff --git a/webrtc/voice_engine/test/auto_test/voe_extended_test.h b/webrtc/voice_engine/test/auto_test/voe_extended_test.h
index f784e9670..403788905 100644
--- a/webrtc/voice_engine/test/auto_test/voe_extended_test.h
+++ b/webrtc/voice_engine/test/auto_test/voe_extended_test.h
@@ -55,7 +55,7 @@ class AudioDeviceModuleImpl : public AudioDeviceModule {
                           uint32_t& position) const {
     return 0;
   }
-  virtual int32_t ChangeUniqueId(const int32_t id) {
+  virtual int32_t ChangeUniqueId(int32_t id) {
     return 0;
   }
   virtual int32_t TimeUntilNextProcess() {
@@ -284,14 +284,14 @@ class AudioDeviceModuleImpl : public AudioDeviceModule {
   virtual int32_t StereoRecording(bool* enabled) const {
     return -1;
   }
-  virtual int32_t SetRecordingChannel(const ChannelType channel) {
+  virtual int32_t SetRecordingChannel(ChannelType channel) {
     return -1;
   }
   virtual int32_t RecordingChannel(ChannelType* channel) const {
     return -1;
   }
 
-  virtual int32_t SetPlayoutBuffer(const BufferType type, uint16_t sizeMS = 0) {
+  virtual int32_t SetPlayoutBuffer(BufferType type, uint16_t sizeMS = 0) {
     return -1;
   }
   virtual int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS) const {
@@ -323,13 +323,13 @@ class AudioDeviceModuleImpl : public AudioDeviceModule {
     return -1;
   }
 
-  virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) {
+  virtual int32_t SetRecordingSampleRate(uint32_t samplesPerSec) {
     return -1;
   }
   virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const {
     return -1;
   }
-  virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) {
+  virtual int32_t SetPlayoutSampleRate(uint32_t samplesPerSec) {
     return -1;
   }
   virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const {
@@ -398,11 +398,11 @@ class XRTPObserver : public VoERTPObserver {
  public:
   XRTPObserver();
   ~XRTPObserver();
-  virtual void OnIncomingCSRCChanged(const int channel,
-                                     const unsigned int CSRC,
-                                     const bool added);
-  virtual void OnIncomingSSRCChanged(const int channel,
-                                     const unsigned int SSRC);
+  virtual void OnIncomingCSRCChanged(int channel,
+                                     unsigned int CSRC,
+                                     bool added);
+  virtual void OnIncomingSSRCChanged(int channel,
+                                     unsigned int SSRC);
  public:
   unsigned int _SSRC;
 };
@@ -442,12 +442,12 @@ class VoEExtendedTest : public VoiceEngineObserver,
 
  protected:
   // from VoiceEngineObserver
-  void CallbackOnError(const int errCode, const int channel);
-  void CallbackOnTrace(const TraceLevel level, const char* message,
-                       const int length);
+  void CallbackOnError(int errCode, int channel);
+  void CallbackOnTrace(TraceLevel level, const char* message,
+                       int length);
 
   // from VoEConnectionObserver
-  void OnPeriodicDeadOrAlive(const int channel, const bool alive);
+  void OnPeriodicDeadOrAlive(int channel, bool alive);
 
  private:
   void Play(int channel, unsigned int timeMillisec,
diff --git a/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc b/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
index 0c734df0c..aed430e74 100644
--- a/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
+++ b/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
@@ -65,10 +65,10 @@ void RunTest(std::string out_path);
 
 class MyObserver : public VoiceEngineObserver {
  public:
-   virtual void CallbackOnError(const int channel, const int err_code);
+   virtual void CallbackOnError(int channel, int err_code);
 };
 
-void MyObserver::CallbackOnError(const int channel, const int err_code) {
+void MyObserver::CallbackOnError(int channel, int err_code) {
   // Add printf for other error codes here
   if (err_code == VE_TYPING_NOISE_WARNING) {
     printf("  TYPING NOISE DETECTED \n");
diff --git a/webrtc/voice_engine/test/win_test/WinTestDlg.cc b/webrtc/voice_engine/test/win_test/WinTestDlg.cc
index fee292065..67da771d4 100644
--- a/webrtc/voice_engine/test/win_test/WinTestDlg.cc
+++ b/webrtc/voice_engine/test/win_test/WinTestDlg.cc
@@ -61,14 +61,14 @@ class ConnectionObserver : public  VoEConnectionObserver
 {
 public:
     ConnectionObserver();
-    virtual void OnPeriodicDeadOrAlive(const int channel, const bool alive);
+    virtual void OnPeriodicDeadOrAlive(int channel, bool alive);
 };
 
 ConnectionObserver::ConnectionObserver()
 {
 }
 
-void ConnectionObserver::OnPeriodicDeadOrAlive(const int channel, const bool alive)
+void ConnectionObserver::OnPeriodicDeadOrAlive(int channel, bool alive)
 {
     CString str;
     str.Format(_T("OnPeriodicDeadOrAlive(channel=%d) => alive=%d"), channel, alive);
@@ -79,7 +79,7 @@ void ConnectionObserver::OnPeriodicDeadOrAlive(const int channel, const bool ali
 //    VoiceEngineObserver
 // ----------------------------------------------------------------------------
 
-void CWinTestDlg::CallbackOnError(const int channel, const int errCode)
+void CWinTestDlg::CallbackOnError(int channel, int errCode)
 {
     _nErrorCallbacks++;
 
@@ -129,14 +129,14 @@ void CWinTestDlg::CallbackOnError(const int channel, const int errCode)
 //    VoERTPObserver
 // ----------------------------------------------------------------------------
 
-void CWinTestDlg::OnIncomingCSRCChanged(const int channel, const unsigned int CSRC, const bool added)
+void CWinTestDlg::OnIncomingCSRCChanged(int channel, unsigned int CSRC, bool added)
 {
     CString str;
     str.Format(_T("OnIncomingCSRCChanged(channel=%d) => CSRC=%u, added=%d"), channel, CSRC, added);
     SetDlgItemText(IDC_EDIT_ERROR_CALLBACK, (LPCTSTR)str);
 }
 
-void CWinTestDlg::OnIncomingSSRCChanged(const int channel, const unsigned int SSRC)
+void CWinTestDlg::OnIncomingSSRCChanged(int channel, unsigned int SSRC)
 {
     CString str;
     str.Format(_T("OnIncomingSSRCChanged(channel=%d) => SSRC=%u"), channel, SSRC);
@@ -184,24 +184,24 @@ class MediaProcessImpl : public VoEMediaProcess
 {
 public:
     MediaProcessImpl();
-    virtual void Process(const int channel,
-                         const ProcessingTypes type,
+    virtual void Process(int channel,
+                         ProcessingTypes type,
                          int16_t audio_10ms[],
-                         const int length,
-                         const int samplingFreqHz,
-                         const bool stereo);
+                         int length,
+                         int samplingFreqHz,
+                         bool stereo);
 };
 
 MediaProcessImpl::MediaProcessImpl()
 {
 }
 
-void MediaProcessImpl::Process(const int channel,
-                               const ProcessingTypes type,
+void MediaProcessImpl::Process(int channel,
+                               ProcessingTypes type,
                                int16_t audio_10ms[],
-                               const int length,
-                               const int samplingFreqHz,
-                               const bool stereo)
+                               int length,
+                               int samplingFreqHz,
+                               bool stereo)
 {
     int x = rand() % 100;
 
diff --git a/webrtc/voice_engine/test/win_test/WinTestDlg.h b/webrtc/voice_engine/test/win_test/WinTestDlg.h
index 412c220cd..89eb8c2c2 100644
--- a/webrtc/voice_engine/test/win_test/WinTestDlg.h
+++ b/webrtc/voice_engine/test/win_test/WinTestDlg.h
@@ -125,13 +125,13 @@ protected:
     virtual void DoDataExchange(CDataExchange* pDX);    // DDX/DDV support
 
 protected:  // VoiceEngineObserver
-    virtual void CallbackOnError(const int channel, const int errCode);
+    virtual void CallbackOnError(int channel, int errCode);
 
 protected:    // VoERTPObserver
     virtual void OnIncomingCSRCChanged(
-        const int channel, const unsigned int CSRC, const bool added);
+        int channel, unsigned int CSRC, bool added);
     virtual void OnIncomingSSRCChanged(
-        const int channel, const unsigned int SSRC);
+        int channel, unsigned int SSRC);
 
 // Implementation
 protected:
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index 3af2603e3..f594ef0e5 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -81,8 +81,8 @@ TransmitMixer::OnPeriodicProcess()
 }
 
 
-void TransmitMixer::PlayNotification(const int32_t id,
-                                     const uint32_t durationMs)
+void TransmitMixer::PlayNotification(int32_t id,
+                                     uint32_t durationMs)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::PlayNotification(id=%d, durationMs=%d)",
@@ -91,8 +91,8 @@ void TransmitMixer::PlayNotification(const int32_t id,
     // Not implement yet
 }
 
-void TransmitMixer::RecordNotification(const int32_t id,
-                                       const uint32_t durationMs)
+void TransmitMixer::RecordNotification(int32_t id,
+                                       uint32_t durationMs)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
                  "TransmitMixer::RecordNotification(id=%d, durationMs=%d)",
@@ -101,7 +101,7 @@ void TransmitMixer::RecordNotification(const int32_t id,
     // Not implement yet
 }
 
-void TransmitMixer::PlayFileEnded(const int32_t id)
+void TransmitMixer::PlayFileEnded(int32_t id)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::PlayFileEnded(id=%d)", id);
@@ -117,7 +117,7 @@ void TransmitMixer::PlayFileEnded(const int32_t id)
 }
 
 void
-TransmitMixer::RecordFileEnded(const int32_t id)
+TransmitMixer::RecordFileEnded(int32_t id)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::RecordFileEnded(id=%d)", id);
@@ -140,7 +140,7 @@ TransmitMixer::RecordFileEnded(const int32_t id)
 }
 
 int32_t
-TransmitMixer::Create(TransmitMixer*& mixer, const uint32_t instanceId)
+TransmitMixer::Create(TransmitMixer*& mixer, uint32_t instanceId)
 {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, -1),
                  "TransmitMixer::Create(instanceId=%d)", instanceId);
@@ -165,7 +165,7 @@ TransmitMixer::Destroy(TransmitMixer*& mixer)
     }
 }
 
-TransmitMixer::TransmitMixer(const uint32_t instanceId) :
+TransmitMixer::TransmitMixer(uint32_t instanceId) :
     _engineStatisticsPtr(NULL),
     _channelManagerPtr(NULL),
     audioproc_(NULL),
@@ -327,13 +327,13 @@ void TransmitMixer::GetSendCodecInfo(int* max_sample_rate, int* max_channels) {
 
 int32_t
 TransmitMixer::PrepareDemux(const void* audioSamples,
-                            const uint32_t nSamples,
-                            const uint8_t nChannels,
-                            const uint32_t samplesPerSec,
-                            const uint16_t totalDelayMS,
-                            const int32_t clockDrift,
-                            const uint16_t currentMicLevel,
-                            const bool keyPressed)
+                            uint32_t nSamples,
+                            uint8_t nChannels,
+                            uint32_t samplesPerSec,
+                            uint16_t totalDelayMS,
+                            int32_t clockDrift,
+                            uint16_t currentMicLevel,
+                            bool keyPressed)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::PrepareDemux(nSamples=%u, nChannels=%u,"
@@ -470,7 +470,7 @@ uint32_t TransmitMixer::CaptureLevel() const
 }
 
 void
-TransmitMixer::UpdateMuteMicrophoneTime(const uint32_t lengthMs)
+TransmitMixer::UpdateMuteMicrophoneTime(uint32_t lengthMs)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
                "TransmitMixer::UpdateMuteMicrophoneTime(lengthMs=%d)",
@@ -488,11 +488,11 @@ TransmitMixer::StopSend()
 }
 
 int TransmitMixer::StartPlayingFileAsMicrophone(const char* fileName,
-                                                const bool loop,
-                                                const FileFormats format,
-                                                const int startPosition,
-                                                const float volumeScaling,
-                                                const int stopPosition,
+                                                bool loop,
+                                                FileFormats format,
+                                                int startPosition,
+                                                float volumeScaling,
+                                                int stopPosition,
                                                 const CodecInst* codecInst)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
@@ -559,10 +559,10 @@ int TransmitMixer::StartPlayingFileAsMicrophone(const char* fileName,
 }
 
 int TransmitMixer::StartPlayingFileAsMicrophone(InStream* stream,
-                                                const FileFormats format,
-                                                const int startPosition,
-                                                const float volumeScaling,
-                                                const int stopPosition,
+                                                FileFormats format,
+                                                int startPosition,
+                                                float volumeScaling,
+                                                int stopPosition,
                                                 const CodecInst* codecInst)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
@@ -671,7 +671,7 @@ int TransmitMixer::IsPlayingFileAsMicrophone() const
     return _filePlaying;
 }
 
-int TransmitMixer::ScaleFileAsMicrophonePlayout(const float scale)
+int TransmitMixer::ScaleFileAsMicrophonePlayout(float scale)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::ScaleFileAsMicrophonePlayout(scale=%5.3f)",
@@ -1208,7 +1208,7 @@ int TransmitMixer::GenerateAudioFrame(const int16_t audio[],
 }
 
 int32_t TransmitMixer::RecordAudioToFile(
-    const uint32_t mixingFrequency)
+    uint32_t mixingFrequency)
 {
     CriticalSectionScoped cs(&_critSect);
     if (_fileRecorderPtr == NULL)
@@ -1231,7 +1231,7 @@ int32_t TransmitMixer::RecordAudioToFile(
 }
 
 int32_t TransmitMixer::MixOrReplaceAudioWithFile(
-    const int mixingFrequency)
+    int mixingFrequency)
 {
     scoped_array<int16_t> fileBuffer(new int16_t[640]);
 
@@ -1328,7 +1328,7 @@ void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift,
 }
 
 #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-int TransmitMixer::TypingDetection(const bool keyPressed)
+int TransmitMixer::TypingDetection(bool keyPressed)
 {
 
     // We let the VAD determine if we're using this feature or not.
diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h
index 44153b156..450f1254b 100644
--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/voice_engine/transmit_mixer.h
@@ -39,7 +39,7 @@ class TransmitMixer : public MonitorObserver,
 
 {
 public:
-    static int32_t Create(TransmitMixer*& mixer, const uint32_t instanceId);
+    static int32_t Create(TransmitMixer*& mixer, uint32_t instanceId);
 
     static void Destroy(TransmitMixer*& mixer);
 
@@ -51,13 +51,13 @@ public:
         AudioProcessing* audioProcessingModule);
 
     int32_t PrepareDemux(const void* audioSamples,
-                         const uint32_t nSamples,
-                         const uint8_t  nChannels,
-                         const uint32_t samplesPerSec,
-                         const uint16_t totalDelayMS,
-                         const int32_t  clockDrift,
-                         const uint16_t currentMicLevel,
-                         const bool keyPressed);
+                         uint32_t nSamples,
+                         uint8_t  nChannels,
+                         uint32_t samplesPerSec,
+                         uint16_t totalDelayMS,
+                         int32_t  clockDrift,
+                         uint16_t currentMicLevel,
+                         bool keyPressed);
 
 
     int32_t DemuxAndMix();
@@ -69,7 +69,7 @@ public:
     int32_t StopSend();
 
     // VoEDtmf
-    void UpdateMuteMicrophoneTime(const uint32_t lengthMs);
+    void UpdateMuteMicrophoneTime(uint32_t lengthMs);
 
     // VoEExternalMedia
     int RegisterExternalMediaProcessing(VoEMediaProcess* object,
@@ -79,7 +79,7 @@ public:
     int GetMixingFrequency();
 
     // VoEVolumeControl
-    int SetMute(const bool enable);
+    int SetMute(bool enable);
 
     bool Mute() const;
 
@@ -92,25 +92,25 @@ public:
     bool IsRecordingMic();
 
     int StartPlayingFileAsMicrophone(const char* fileName,
-                                     const bool loop,
-                                     const FileFormats format,
-                                     const int startPosition,
-                                     const float volumeScaling,
-                                     const int stopPosition,
+                                     bool loop,
+                                     FileFormats format,
+                                     int startPosition,
+                                     float volumeScaling,
+                                     int stopPosition,
                                      const CodecInst* codecInst);
 
     int StartPlayingFileAsMicrophone(InStream* stream,
-                                     const FileFormats format,
-                                     const int startPosition,
-                                     const float volumeScaling,
-                                     const int stopPosition,
+                                     FileFormats format,
+                                     int startPosition,
+                                     float volumeScaling,
+                                     int stopPosition,
                                      const CodecInst* codecInst);
 
     int StopPlayingFileAsMicrophone();
 
     int IsPlayingFileAsMicrophone() const;
 
-    int ScaleFileAsMicrophonePlayout(const float scale);
+    int ScaleFileAsMicrophonePlayout(float scale);
 
     int StartRecordingMicrophone(const char* fileName,
                                  const CodecInst* codecInst);
@@ -137,15 +137,15 @@ public:
 
 
     // FileCallback
-    void PlayNotification(const int32_t id,
-                          const uint32_t durationMs);
+    void PlayNotification(int32_t id,
+                          uint32_t durationMs);
 
-    void RecordNotification(const int32_t id,
-                            const uint32_t durationMs);
+    void RecordNotification(int32_t id,
+                            uint32_t durationMs);
 
-    void PlayFileEnded(const int32_t id);
+    void PlayFileEnded(int32_t id);
 
-    void RecordFileEnded(const int32_t id);
+    void RecordFileEnded(int32_t id);
 
 #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
     // Typing detection
@@ -161,7 +161,7 @@ public:
   bool IsStereoChannelSwappingEnabled();
 
 private:
-    TransmitMixer(const uint32_t instanceId);
+    TransmitMixer(uint32_t instanceId);
 
     // Gets the maximum sample rate and number of channels over all currently
     // sending codecs.
@@ -171,15 +171,15 @@ private:
                            int nSamples,
                            int nChannels,
                            int samplesPerSec);
-    int32_t RecordAudioToFile(const uint32_t mixingFrequency);
+    int32_t RecordAudioToFile(uint32_t mixingFrequency);
 
     int32_t MixOrReplaceAudioWithFile(
-        const int mixingFrequency);
+        int mixingFrequency);
 
     void ProcessAudio(int delay_ms, int clock_drift, int current_mic_level);
 
 #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-    int TypingDetection(const bool keyPressed);
+    int TypingDetection(bool keyPressed);
 #endif
 
     // uses
diff --git a/webrtc/voice_engine/transmit_mixer_unittest.cc b/webrtc/voice_engine/transmit_mixer_unittest.cc
index d8d85b62b..8b28ec494 100644
--- a/webrtc/voice_engine/transmit_mixer_unittest.cc
+++ b/webrtc/voice_engine/transmit_mixer_unittest.cc
@@ -19,9 +19,9 @@ namespace {
 
 class MediaCallback : public VoEMediaProcess {
  public:
-  virtual void Process(const int channel, const ProcessingTypes type,
-                       int16_t audio[], const int samples_per_channel,
-                       const int sample_rate_hz, const bool is_stereo) {
+  virtual void Process(int channel, ProcessingTypes type,
+                       int16_t audio[], int samples_per_channel,
+                       int sample_rate_hz, bool is_stereo) {
   }
 };
 
diff --git a/webrtc/voice_engine/voe_audio_processing_impl.cc b/webrtc/voice_engine/voe_audio_processing_impl.cc
index ac491db60..71a1904f5 100644
--- a/webrtc/voice_engine/voe_audio_processing_impl.cc
+++ b/webrtc/voice_engine/voe_audio_processing_impl.cc
@@ -263,7 +263,7 @@ int VoEAudioProcessingImpl::GetAgcStatus(bool& enabled, AgcModes& mode) {
 #endif
 }
 
-int VoEAudioProcessingImpl::SetAgcConfig(const AgcConfig config) {
+int VoEAudioProcessingImpl::SetAgcConfig(AgcConfig config) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetAgcConfig()");
 #ifdef WEBRTC_VOICE_ENGINE_AGC
@@ -437,7 +437,7 @@ int VoEAudioProcessingImpl::GetRxAgcStatus(int channel,
 }
 
 int VoEAudioProcessingImpl::SetRxAgcConfig(int channel,
-                                           const AgcConfig config) {
+                                           AgcConfig config) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetRxAgcConfig(channel=%d)", channel);
 #ifdef WEBRTC_VOICE_ENGINE_AGC
diff --git a/webrtc/voice_engine/voe_audio_processing_impl.h b/webrtc/voice_engine/voe_audio_processing_impl.h
index 3d6b64de7..8686b3cf5 100644
--- a/webrtc/voice_engine/voe_audio_processing_impl.h
+++ b/webrtc/voice_engine/voe_audio_processing_impl.h
@@ -27,7 +27,7 @@ class VoEAudioProcessingImpl : public VoEAudioProcessing {
 
   virtual int GetAgcStatus(bool& enabled, AgcModes& mode);
 
-  virtual int SetAgcConfig(const AgcConfig config);
+  virtual int SetAgcConfig(AgcConfig config);
 
   virtual int GetAgcConfig(AgcConfig& config);
 
@@ -43,7 +43,7 @@ class VoEAudioProcessingImpl : public VoEAudioProcessing {
 
   virtual int GetRxAgcStatus(int channel, bool& enabled, AgcModes& mode);
 
-  virtual int SetRxAgcConfig(int channel, const AgcConfig config);
+  virtual int SetRxAgcConfig(int channel, AgcConfig config);
 
   virtual int GetRxAgcConfig(int channel, AgcConfig& config);
 
diff --git a/webrtc/voice_engine/voe_base_impl.cc b/webrtc/voice_engine/voe_base_impl.cc
index aeec9bcc9..1b14be72f 100644
--- a/webrtc/voice_engine/voe_base_impl.cc
+++ b/webrtc/voice_engine/voe_base_impl.cc
@@ -65,7 +65,7 @@ VoEBaseImpl::~VoEBaseImpl()
     delete &_callbackCritSect;
 }
 
-void VoEBaseImpl::OnErrorIsReported(const ErrorCode error)
+void VoEBaseImpl::OnErrorIsReported(ErrorCode error)
 {
     CriticalSectionScoped cs(&_callbackCritSect);
     if (_voiceEngineObserver)
@@ -94,7 +94,7 @@ void VoEBaseImpl::OnErrorIsReported(const ErrorCode error)
     }
 }
 
-void VoEBaseImpl::OnWarningIsReported(const WarningCode warning)
+void VoEBaseImpl::OnWarningIsReported(WarningCode warning)
 {
     CriticalSectionScoped cs(&_callbackCritSect);
     if (_voiceEngineObserver)
@@ -126,14 +126,14 @@ void VoEBaseImpl::OnWarningIsReported(const WarningCode warning)
 
 int32_t VoEBaseImpl::RecordedDataIsAvailable(
         const void* audioSamples,
-        const uint32_t nSamples,
-        const uint8_t nBytesPerSample,
-        const uint8_t nChannels,
-        const uint32_t samplesPerSec,
-        const uint32_t totalDelayMS,
-        const int32_t clockDrift,
-        const uint32_t currentMicLevel,
-        const bool keyPressed,
+        uint32_t nSamples,
+        uint8_t nBytesPerSample,
+        uint8_t nChannels,
+        uint32_t samplesPerSec,
+        uint32_t totalDelayMS,
+        int32_t clockDrift,
+        uint32_t currentMicLevel,
+        bool keyPressed,
         uint32_t& newMicLevel)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1),
@@ -232,10 +232,10 @@ int32_t VoEBaseImpl::RecordedDataIsAvailable(
 }
 
 int32_t VoEBaseImpl::NeedMorePlayData(
-        const uint32_t nSamples,
-        const uint8_t nBytesPerSample,
-        const uint8_t nChannels,
-        const uint32_t samplesPerSec,
+        uint32_t nSamples,
+        uint8_t nBytesPerSample,
+        uint8_t nChannels,
+        uint32_t samplesPerSec,
         void* audioSamples,
         uint32_t& nSamplesOut)
 {
diff --git a/webrtc/voice_engine/voe_base_impl.h b/webrtc/voice_engine/voe_base_impl.h
index 9d89733e3..230e750e1 100644
--- a/webrtc/voice_engine/voe_base_impl.h
+++ b/webrtc/voice_engine/voe_base_impl.h
@@ -73,26 +73,26 @@ public:
     // AudioTransport
     virtual int32_t
         RecordedDataIsAvailable(const void* audioSamples,
-                                const uint32_t nSamples,
-                                const uint8_t nBytesPerSample,
-                                const uint8_t nChannels,
-                                const uint32_t samplesPerSec,
-                                const uint32_t totalDelayMS,
-                                const int32_t clockDrift,
-                                const uint32_t currentMicLevel,
-                                const bool keyPressed,
+                                uint32_t nSamples,
+                                uint8_t nBytesPerSample,
+                                uint8_t nChannels,
+                                uint32_t samplesPerSec,
+                                uint32_t totalDelayMS,
+                                int32_t clockDrift,
+                                uint32_t currentMicLevel,
+                                bool keyPressed,
                                 uint32_t& newMicLevel);
 
-    virtual int32_t NeedMorePlayData(const uint32_t nSamples,
-                                     const uint8_t nBytesPerSample,
-                                     const uint8_t nChannels,
-                                     const uint32_t samplesPerSec,
+    virtual int32_t NeedMorePlayData(uint32_t nSamples,
+                                     uint8_t nBytesPerSample,
+                                     uint8_t nChannels,
+                                     uint32_t samplesPerSec,
                                      void* audioSamples,
                                      uint32_t& nSamplesOut);
 
     // AudioDeviceObserver
-    virtual void OnErrorIsReported(const ErrorCode error);
-    virtual void OnWarningIsReported(const WarningCode warning);
+    virtual void OnErrorIsReported(ErrorCode error);
+    virtual void OnWarningIsReported(WarningCode warning);
 
 protected:
     VoEBaseImpl(voe::SharedData* shared);
diff --git a/webrtc/voice_engine/voe_rtp_rtcp_impl.cc b/webrtc/voice_engine/voe_rtp_rtcp_impl.cc
index 3470443c2..6999d51cc 100644
--- a/webrtc/voice_engine/voe_rtp_rtcp_impl.cc
+++ b/webrtc/voice_engine/voe_rtp_rtcp_impl.cc
@@ -403,7 +403,7 @@ int VoERTP_RTCPImpl::GetRemoteRTCPData(
 
 int VoERTP_RTCPImpl::SendApplicationDefinedRTCPPacket(
     int channel,
-    const unsigned char subType,
+    unsigned char subType,
     unsigned int name,
     const char* data,
     unsigned short dataLengthInBytes)
diff --git a/webrtc/voice_engine/voe_rtp_rtcp_impl.h b/webrtc/voice_engine/voe_rtp_rtcp_impl.h
index 3d64205b9..9c0392b6d 100644
--- a/webrtc/voice_engine/voe_rtp_rtcp_impl.h
+++ b/webrtc/voice_engine/voe_rtp_rtcp_impl.h
@@ -50,7 +50,7 @@ public:
 
     virtual int SendApplicationDefinedRTCPPacket(
         int channel,
-        const unsigned char subType,
+        unsigned char subType,
         unsigned int name,
         const char* data,
         unsigned short dataLengthInBytes);
diff --git a/webrtc/voice_engine/voice_engine_defines.h b/webrtc/voice_engine/voice_engine_defines.h
index 18600e116..cb63940df 100644
--- a/webrtc/voice_engine/voice_engine_defines.h
+++ b/webrtc/voice_engine/voice_engine_defines.h
@@ -197,7 +197,7 @@ enum { kVoiceEngineMaxRtpExtensionId = 14 };
 namespace webrtc
 {
 
-inline int VoEId(const int veId, const int chId)
+inline int VoEId(int veId, int chId)
 {
     if (chId == -1)
     {
@@ -207,13 +207,13 @@ inline int VoEId(const int veId, const int chId)
     return (int) ((veId << 16) + chId);
 }
 
-inline int VoEModuleId(const int veId, const int chId)
+inline int VoEModuleId(int veId, int chId)
 {
     return (int) ((veId << 16) + chId);
 }
 
 // Convert module ID to internal VoE channel ID
-inline int VoEChannelId(const int moduleId)
+inline int VoEChannelId(int moduleId)
 {
     return (int) (moduleId & 0xffff);
 }
diff --git a/webrtc/voice_engine/voice_engine_impl.cc b/webrtc/voice_engine/voice_engine_impl.cc
index fd8723a67..30609b26f 100644
--- a/webrtc/voice_engine/voice_engine_impl.cc
+++ b/webrtc/voice_engine/voice_engine_impl.cc
@@ -80,7 +80,7 @@ VoiceEngine* VoiceEngine::Create()
     return GetVoiceEngine();
 }
 
-int VoiceEngine::SetTraceFilter(const unsigned int filter)
+int VoiceEngine::SetTraceFilter(unsigned int filter)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
                  VoEId(gVoiceEngineInstanceCounter, -1),
@@ -104,7 +104,7 @@ int VoiceEngine::SetTraceFilter(const unsigned int filter)
 }
 
 int VoiceEngine::SetTraceFile(const char* fileNameUTF8,
-                              const bool addFileCounter)
+                              bool addFileCounter)
 {
     int ret = Trace::SetTraceFile(fileNameUTF8, addFileCounter);
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice,