Review URL: http://webrtc-codereview.appspot.com/109006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@383 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/rtp_rtcp/source/forward_error_correction.cc

@@ -9,7 +9,6 @@
  */
 
 #include "forward_error_correction.h"
-#include "fec_private_tables.h"
 #include "rtp_utility.h"
 
 #include "trace.h"

src/modules/rtp_rtcp/source/remote_rate_control.cc

@@ -236,10 +236,12 @@ WebRtc_UWord32 RemoteRateControl::ChangeBitRate(WebRtc_UWord32 currentBitRate,
 #ifdef _DEBUG
             //char logStr[256];
 #ifdef _WIN32
-            _snprintf(logStr,256, "New bitRate: %lu\n", currentBitRate / 1000);
+            _snprintf(logStr,256, "New bitRate: %lu\n",
+                      static_cast<long unsigned int> (currentBitRate / 1000));
             OutputDebugStringA(logStr);
 #else
-            snprintf(logStr,256, "New bitRate: %lu\n", currentBitRate / 1000);
+            snprintf(logStr,256, "New bitRate: %lu\n",
+                     static_cast<long unsigned int> (currentBitRate / 1000));
             //TODO
 #endif
 #endif

src/modules/rtp_rtcp/source/rtp_receiver_video.cc

@@ -35,11 +35,11 @@ RTPReceiverVideo::RTPReceiverVideo(const WebRtc_Word32 id,
     _criticalSectionReceiverVideo(*CriticalSectionWrapper::CreateCriticalSection()),
 
     _completeFrame(false),
-    _receiveFEC(NULL),
     _packetStartTimeMs(0),
     _receivedBW(),
     _estimatedBW(0),
     _currentFecFrameDecoded(false),
+    _receiveFEC(NULL),
     _h263InverseLogic(false),
     _overUseDetector(),
     _videoBitRate(),

src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -2187,10 +2187,10 @@ ModuleRtpRtcpImpl::OnBandwidthEstimateUpdate(WebRtc_UWord16 bandWidthKbit)
             extensions.  Overhead excludes any RTP payload headers and the
             payload itself.
         */
-        WebRtc_UWord16 RTPpacketOH = _rtpReceiver.PacketOHReceived();
+        _rtpReceiver.PacketOHReceived();
 
         // call RequestTMMBR when our localy created estimate changes
-        _rtcpSender.RequestTMMBR(bandWidthKbit, 0/*RTPpacketOH + _packetOverHead*/);
+        _rtcpSender.RequestTMMBR(bandWidthKbit, 0);
     }
 }
 

src/modules/rtp_rtcp/source/rtp_sender.cc

@@ -611,9 +611,9 @@ RTPSender::CheckPayloadType(const WebRtc_Word8 payloadType,
                         _audio->SetAudioFrequency(payloadFreqHz);
                         // We need to correct the timestamp again,
                         // since this might happen after we've set it
-                        WebRtc_UWord32 RTPtime = 
+                        WebRtc_UWord32 RTPtime =
                             ModuleRTPUtility::CurrentRTP(payloadFreqHz);
-                        SetStartTimestamp(RTPtime); 
+                        SetStartTimestamp(RTPtime);
                         // will be ignored if it's already configured via API
                     }
                 }

src/modules/rtp_rtcp/source/rtp_sender_audio.cc

@@ -87,7 +87,7 @@ RTPSenderAudio::SetAudioFrequency(const WebRtc_UWord32 f)
     _frequency = f;
 }
 
-WebRtc_UWord32
+int
 RTPSenderAudio::AudioFrequency() const
 {
     CriticalSectionScoped cs(_sendAudioCritsect);

src/modules/rtp_rtcp/source/rtp_sender_audio.h

@@ -71,7 +71,7 @@ public:
 
     void SetAudioFrequency(const WebRtc_UWord32 f);
 
-    WebRtc_UWord32 AudioFrequency() const;
+    int AudioFrequency() const;
 
     // Set payload type for Redundant Audio Data RFC 2198
     WebRtc_Word32 SetRED(const WebRtc_Word8 payloadType);

src/modules/rtp_rtcp/source/rtp_sender_video.cc

@@ -170,7 +170,7 @@ RTPSenderVideo::SendVideoPacket(const FrameType frameType,
         // Add packet to FEC list
         _rtpPacketListFec.PushBack(ptrGenericFEC);
         // FEC can only protect up to kMaxMediaPackets packets
-        if (_mediaPacketListFec.GetSize() <
+        if (static_cast<int>(_mediaPacketListFec.GetSize()) <
             ForwardErrorCorrection::kMaxMediaPackets)
         {
             _mediaPacketListFec.PushBack(ptrGenericFEC->pkt);
@@ -195,7 +195,7 @@ RTPSenderVideo::SendVideoPacket(const FrameType frameType,
 
             // Number of first partition packets cannot exceed kMaxMediaPackets
             if (_numberFirstPartition >
-                    ForwardErrorCorrection::kMaxMediaPackets)
+                ForwardErrorCorrection::kMaxMediaPackets)
             {
                 _numberFirstPartition =
                     ForwardErrorCorrection::kMaxMediaPackets;

src/modules/rtp_rtcp/source/rtp_sender_video.h

@@ -160,7 +160,7 @@ private:
     bool                      _useUepProtectionDelta;
     WebRtc_UWord8             _fecProtectionFactor;
     bool                      _fecUseUepProtection;
-    WebRtc_UWord32            _numberFirstPartition;
+    int                       _numberFirstPartition;
     ListWrapper               _mediaPacketListFec;
     ListWrapper               _rtpPacketListFec;
 

src/modules/rtp_rtcp/source/rtp_utility.cc

@@ -567,11 +567,11 @@ ModuleRTPUtility::RTPHeaderParser::Parse(WebRtcRTPHeader& parsedPacket) const
             */
 
             // Parse out the fields but only use it for debugging for now.
-            const WebRtc_UWord8 ID = (*ptr & 0xf0) >> 4;
+            //const WebRtc_UWord8 ID = (*ptr & 0xf0) >> 4;
-            const WebRtc_UWord8 len = (*ptr & 0x0f);
+            //const WebRtc_UWord8 len = (*ptr & 0x0f);
             ptr++;
-            const WebRtc_UWord8 V = (*ptr & 0x80) >> 7;
+            //const WebRtc_UWord8 V = (*ptr & 0x80) >> 7;
-            const WebRtc_UWord8 level = (*ptr & 0x7f);
+            //const WebRtc_UWord8 level = (*ptr & 0x7f);
             // DEBUG_PRINT("RTP_AUDIO_LEVEL_UNIQUE_ID: ID=%u, len=%u, V=%u, level=%u", ID, len, V, level);
         }
         parsedPacket.header.headerLength += XLen;