diff --git a/src/modules/video_coding/codecs/test_framework/normal_test.cc b/src/modules/video_coding/codecs/test_framework/normal_test.cc
index 0cc20efb8e..2dd7bce2fd 100644
--- a/src/modules/video_coding/codecs/test_framework/normal_test.cc
+++ b/src/modules/video_coding/codecs/test_framework/normal_test.cc
@@ -235,7 +235,7 @@ NormalTest::Encode()
 }
 
 int
-NormalTest::Decode()
+NormalTest::Decode(int lossValue)
 {
     _encodedVideoBuffer.SetWidth(_inst.width);
     _encodedVideoBuffer.SetHeight(_inst.height);
diff --git a/src/modules/video_coding/codecs/test_framework/normal_test.h b/src/modules/video_coding/codecs/test_framework/normal_test.h
index c7bb676f62..0dcd9c269e 100644
--- a/src/modules/video_coding/codecs/test_framework/normal_test.h
+++ b/src/modules/video_coding/codecs/test_framework/normal_test.h
@@ -26,7 +26,7 @@ protected:
     virtual void Setup();
     virtual void Teardown();
     virtual bool Encode();
-    virtual int Decode();
+    virtual int Decode(int lossValue = 0);
     virtual void CodecSpecific_InitBitrate()=0;
     virtual int DoPacketLoss() {return 0;};
 
diff --git a/src/modules/video_coding/main/test/generic_codec_test.h b/src/modules/video_coding/main/test/generic_codec_test.h
index f8eff1aa36..f7217007f5 100644
--- a/src/modules/video_coding/main/test/generic_codec_test.h
+++ b/src/modules/video_coding/main/test/generic_codec_test.h
@@ -26,6 +26,9 @@ Test consists of:
 4. Decoder control test / General API functionality
 
 */
+
+namespace webrtc {
+
 int VCMGenericCodecTest(CmdArgs& args);
 
 class GenericCodecTest
@@ -97,4 +100,6 @@ private:
     WebRtc_UWord32 _timeStamp;
 }; // end of VCMEncodeCompleteCallback
 
+}  // namespace webrtc
+
 #endif // WEBRTC_MODULES_VIDEO_CODING_TEST_GENERIC_CODEC_TEST_H_
diff --git a/src/modules/video_coding/main/test/media_opt_test.h b/src/modules/video_coding/main/test/media_opt_test.h
index e2bc6f8d04..6f1981cd0f 100644
--- a/src/modules/video_coding/main/test/media_opt_test.h
+++ b/src/modules/video_coding/main/test/media_opt_test.h
@@ -20,8 +20,6 @@
 #include "video_coding.h"
 #include "video_source.h"
 
-using namespace std;
-
 // media optimization test
 // This test simulates a complete encode-decode cycle via the RTP module.
 // allows error resilience tests, packet loss tests, etc.
@@ -76,7 +74,7 @@ private:
     WebRtc_Word32                    _frameCnt;
     float                            _sumEncBytes;
     WebRtc_Word32                    _numFramesDropped;
-    string                           _codecName;
+    std::string                      _codecName;
     webrtc::VideoCodecType           _sendCodecType;
     WebRtc_Word32                    _numberOfCores;
 
diff --git a/src/modules/video_coding/main/test/mt_test_common.cc b/src/modules/video_coding/main/test/mt_test_common.cc
index 0af731c301..7addb876b9 100644
--- a/src/modules/video_coding/main/test/mt_test_common.cc
+++ b/src/modules/video_coding/main/test/mt_test_common.cc
@@ -14,6 +14,7 @@
 
 #include "rtp_dump.h"
 
+namespace webrtc {
 
 TransportCallback::TransportCallback(webrtc::RtpRtcp* rtp,
                                      const char* filename):
@@ -130,3 +131,5 @@ bool TransportThread(void *obj)
     state->_transport.TransportPackets();
     return true;
 }
+
+}  // namespace webrtc
diff --git a/src/modules/video_coding/main/test/mt_test_common.h b/src/modules/video_coding/main/test/mt_test_common.h
index 6dcbceda95..5863d048ca 100644
--- a/src/modules/video_coding/main/test/mt_test_common.h
+++ b/src/modules/video_coding/main/test/mt_test_common.h
@@ -20,7 +20,7 @@
 #include "test_util.h"
 #include "video_coding.h"
 
-using namespace webrtc;
+namespace webrtc {
 
 class SendSharedState
 {
@@ -83,5 +83,6 @@ bool VCMProcessingThread(void* obj);
 bool VCMDecodeThread(void* obj);
 bool TransportThread(void *obj);
 
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_MT_TEST_COMMON_H_
diff --git a/src/modules/video_coding/main/test/test_callbacks.cc b/src/modules/video_coding/main/test/test_callbacks.cc
index 4a4fe004bf..0dd9be3cdd 100644
--- a/src/modules/video_coding/main/test/test_callbacks.cc
+++ b/src/modules/video_coding/main/test/test_callbacks.cc
@@ -15,6 +15,8 @@
 #include "rtp_dump.h"
 #include "test_macros.h"
 
+namespace webrtc {
+
 /******************************
  *  VCMEncodeCompleteCallback
  *****************************/
@@ -503,14 +505,13 @@ VideoProtectionCallback::FECKeyUepProtection()
 
 void
 RTPFeedbackCallback::OnNetworkChanged(const WebRtc_Word32 id,
-                                      const WebRtc_UWord16 bitrateTargetKbit,
+                                      const WebRtc_UWord32 bitrateBps,
                                       const WebRtc_UWord8 fractionLost,
-                                      const WebRtc_UWord16 roundTripTimeMs,
-                                      const WebRtc_UWord32 jitterMS,
-                                      const WebRtc_UWord16 bwEstimateKbitMin,
-                                      const WebRtc_UWord16 bwEstimateKbitMax)
+                                      const WebRtc_UWord16 roundTripTimeMs)
 {
 
-    _vcm->SetChannelParameters(bitrateTargetKbit, fractionLost,
+    _vcm->SetChannelParameters(bitrateBps / 1000, fractionLost,
                                (WebRtc_UWord8)roundTripTimeMs);
 }
+
+}  // namespace webrtc
diff --git a/src/modules/video_coding/main/test/test_callbacks.h b/src/modules/video_coding/main/test/test_callbacks.h
index 637ab96798..deb24b4bc3 100644
--- a/src/modules/video_coding/main/test/test_callbacks.h
+++ b/src/modules/video_coding/main/test/test_callbacks.h
@@ -28,12 +28,9 @@
 #include "trace.h"
 #include "video_coding.h"
 
-using namespace webrtc;
-
 namespace webrtc
 {
     class RtpDump;
-}
 
 // Send Side - Packetization callback - send an encoded frame to the VCMReceiver
 class VCMEncodeCompleteCallback: public VCMPacketizationCallback
@@ -238,7 +235,7 @@ class VideoProtectionCallback: public VCMProtectionCallback
 public:
     VideoProtectionCallback();
     virtual ~VideoProtectionCallback();
-    void RegisterRtpModule(RtpRtcp*  rtp){_rtp = rtp;}
+    void RegisterRtpModule(RtpRtcp* rtp) {_rtp = rtp;}
     WebRtc_Word32 ProtectionRequest(const WebRtc_UWord8 deltaFECRate,
                                     const WebRtc_UWord8 keyFECRate,
                                     const bool deltaUseUepProtection,
@@ -259,23 +256,22 @@ private:
 };
 
 // Feed back from the RTP Module callback
-class RTPFeedbackCallback: public RtpVideoFeedback
-{
-public:
-    RTPFeedbackCallback(VideoCodingModule* vcm) {_vcm = vcm;};
-   void OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
-                                    const WebRtc_UWord8 message = 0){};
+class RTPFeedbackCallback : public RtpVideoFeedback {
+ public:
+  RTPFeedbackCallback(VideoCodingModule* vcm) {_vcm = vcm;};
+  void OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
+                                   const FrameType type,
+                                   const WebRtc_UWord8 streamIdx) {};
 
    void OnNetworkChanged(const WebRtc_Word32 id,
-                          const WebRtc_UWord16 bitrateTargetKbit,
-                          const WebRtc_UWord8 fractionLost,
-                          const WebRtc_UWord16 roundTripTimeMs,
-                          const WebRtc_UWord32 jitterMS,
-                          const WebRtc_UWord16 bwEstimateKbitMin,
-                          const WebRtc_UWord16 bwEstimateKbitMax);
-private:
-   VideoCodingModule* _vcm;
+                         const WebRtc_UWord32 bitrateBps,
+                         const WebRtc_UWord8 fractionLost,
+                         const WebRtc_UWord16 roundTripTimeMs);
+
+ private:
+  VideoCodingModule* _vcm;
 };
 
+}  // namespace webrtc
 
 #endif