Fix compile warnings in Release configuration.

Review URL: http://webrtc-codereview.appspot.com/119003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@424 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/audio_coding/main/source/audio_coding_module_impl.cc

@@ -322,35 +322,34 @@ AudioCodingModuleImpl::TimeUntilNextProcess()
 }
 
 // Process any pending tasks such as timeouts
-WebRtc_Word32 
+WebRtc_Word32
 AudioCodingModuleImpl::Process()
 {
     WebRtc_UWord8 bitStream[2 * MAX_PAYLOAD_SIZE_BYTE]; // Make room for 1 RED payload
     WebRtc_Word16 lengthBytes = 2 * MAX_PAYLOAD_SIZE_BYTE;
+    WebRtc_Word16 redLengthBytes = lengthBytes;
     WebRtc_UWord32 rtpTimestamp;
     WebRtc_Word16 status;
     WebRtcACMEncodingType encodingType;
     FrameType frameType = kAudioFrameSpeech;
-    WebRtc_Word16 redLengthBytes;
     WebRtc_UWord8 currentPayloadType;
     bool hasDataToSend = false;
     bool fecActive = false;
-    WebRtc_UWord32 dummyFragLength;
+
- 
     // keep the scope of the ACM critical section limited
     {
         CriticalSectionScoped lock(*_acmCritSect);
         if(!HaveValidEncoder("Process"))
-        {           
+        {
             return -1;
         }
- 
+
-       status = _codecs[_currentSendCodecIdx]->Encode(bitStream, &lengthBytes, 
+        status = _codecs[_currentSendCodecIdx]->Encode(bitStream, &lengthBytes,
-                &rtpTimestamp, &encodingType);
+                 &rtpTimestamp, &encodingType);
         if (status < 0) // Encode failed
         {
             // logging error
-            WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id, 
+            WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
                 "Process(): Encoding Failed");
             lengthBytes = 0;
             return -1;
@@ -415,7 +414,6 @@ AudioCodingModuleImpl::Process()
             // one RTP packet and the fragmentation points
             // are set.
             // Only apply RED on speech data.
-            
             if((_fecEnabled) &&
                 ((encodingType == kActiveNormalEncoded) ||
                  (encodingType == kPassiveNormalEncoded)))
@@ -481,12 +479,12 @@ AudioCodingModuleImpl::Process()
 
                 // Insert new packet length.
                 _fragmentation->fragmentationLength[0] = lengthBytes;
-                
+
                 // Insert new packet payload type.
                 _fragmentation->fragmentationPlType[0] = currentPayloadType;
                 _lastFECTimestamp = rtpTimestamp;
 
-                // can be modified by the getRedPayload() call if iSAC is utilized
+                // can be modified by the GetRedPayload() call if iSAC is utilized
                 redLengthBytes = lengthBytes;
                 // A fragmentation header is provided => packetization according to RFC 2198
                 // (RTP Payload for Redundant Audio Data) will be used.
@@ -505,11 +503,8 @@ AudioCodingModuleImpl::Process()
                 {
                     // The codec was not iSAC => use current encoder output as redundant data
                     // instead (trivial FEC scheme)
-                    memcpy(_redBuffer, bitStream, redLengthBytes); 
+                    memcpy(_redBuffer, bitStream, redLengthBytes);
                 }
-                
-                // Temporary storing RED length
-                dummyFragLength = redLengthBytes;
 
                 _isFirstRED = false;
                 // Update payload type with RED payload type
@@ -533,10 +528,10 @@ AudioCodingModuleImpl::Process()
         if(_packetizationCallback != NULL)
         {
             if (fecActive) {
-                _packetizationCallback->SendData(frameType, currentPayloadType, 
+                _packetizationCallback->SendData(frameType, currentPayloadType,
                     rtpTimestamp, bitStream, lengthBytes, _fragmentation);
             } else {
-                _packetizationCallback->SendData(frameType, currentPayloadType, 
+                _packetizationCallback->SendData(frameType, currentPayloadType,
                     rtpTimestamp, bitStream, lengthBytes, NULL);
             }
         }
@@ -548,7 +543,7 @@ AudioCodingModuleImpl::Process()
         }
     }
     if (fecActive) {
-        _fragmentation->fragmentationLength[1] = dummyFragLength;
+        _fragmentation->fragmentationLength[1] = redLengthBytes;
     }
     return lengthBytes;
 }

src/video_engine/main/source/vie_file_recorder.cc

@@ -133,7 +133,7 @@ int ViEFileRecorder::StartRecording(const char* fileNameUTF8,
 int ViEFileRecorder::StopRecording()
 {
 
-    int error;
+    int error = 0;
     // Stop recording audio
     // Note - we can not hold the _ptrCritSect while accessing VE functions. It might cause deadlock in Write
     if (_veFileInterface)

src/video_engine/main/source/vie_sender.cc

@@ -304,8 +304,7 @@ int ViESender::SendPacket(int vieId, const void *data, int len)
         return -1;
     }
 
-    int channelId = ChannelId(vieId);
+    assert(ChannelId(vieId) == _channelId);
-    assert(channelId == _channelId);
 
     // Prepare for possible encryption and sending
     WebRtc_UWord8* sendPacket = (WebRtc_UWord8*) data;
@@ -356,8 +355,8 @@ int ViESender::SendRTCPPacket(int vieId, const void *data, int len)
         // No transport
         return -1;
     }
-    int channelId = ChannelId(vieId);
+
-    assert(channelId == _channelId);
+    assert(ChannelId(vieId) == _channelId);
 
     // Prepare for possible encryption and sending
     WebRtc_UWord8* sendPacket = (WebRtc_UWord8*) data;

src/video_engine/main/test/AutoTest/source/vie_autotest_main.cc

@@ -90,7 +90,13 @@ bool ViEAutoTestMain::BeginOSIndependentTesting()
         }
         else
         {
-            scanf("%d", &testType);
+            if (scanf("%d", &testType) <= 0)
+            {
+                ViETest::Log("ERROR: unable to read selection. Try again\n");
+                testType = -1;
+                getchar();
+                continue;
+            }
             getchar();
         }
         ViETest::Log("");

src/voice_engine/main/source/channel.cc

@@ -639,8 +639,7 @@ Channel::OnInitializeDecoder(
                  "payloadName=%s, frequency=%u, channels=%u, rate=%u)",
                  id, payloadType, payloadName, frequency, channels, rate);
 
-    WebRtc_Word32 channel = VoEChannelId(id);
+    assert(VoEChannelId(id) == _channelId);
-    assert(channel == _channelId);
 
     CodecInst receiveCodec;
     CodecInst dummyCodec;
@@ -658,7 +657,7 @@ Channel::OnInitializeDecoder(
     if (_audioCodingModule.RegisterReceiveCodec(receiveCodec) == -1)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                     VoEId(_instanceId,_channelId),
+                     VoEId(_instanceId, _channelId),
                      "Channel::OnInitializeDecoder() invalid codec ("
                      "pt=%d, name=%s) received - 1", payloadType, payloadName);
         _engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR);
@@ -703,8 +702,7 @@ Channel::OnReceivedPacket(const WebRtc_Word32 id,
                  "Channel::OnReceivedPacket(id=%d, packetType=%d)",
                  id, packetType);
 
-    WebRtc_Word32 channel = VoEChannelId(id);
+    assert(VoEChannelId(id) == _channelId);
-    assert(channel == _channelId);
 
     // Notify only for the case when we have restarted an RTP session.
     if (_rtpPacketTimedOut && (kPacketRtp == packetType))