Valgrind reports a racing condition on _sending because it is accessed by

both TransmitMixer::PrepareDemux() and StartSend()/StopSend().
Put a lock to resolve it.
Review URL: http://webrtc-codereview.appspot.com/293005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1038 4adac7df-926f-26a2-2b94-8c16560cd09d

src/voice_engine/main/source/channel.cc

@@ -1694,18 +1694,28 @@ Channel::StartSend()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::StartSend()");
+    {
+        // A lock is needed because |_sending| can be accessed or modified by
+        // another thread at the same time.
+        CriticalSectionScoped cs(_callbackCritSect);
+
         if (_sending)
         {
             return 0;
         }
+        _sending = true;
+    }
+
     if (_rtpRtcpModule.SetSendingStatus(true) != 0)
     {
         _engineStatisticsPtr->SetLastError(
             VE_RTP_RTCP_MODULE_ERROR, kTraceError,
             "StartSend() RTP/RTCP failed to start sending");
+        CriticalSectionScoped cs(_callbackCritSect);
+        _sending = false;
         return -1;
     }
-    _sending = true;
+
     return 0;
 }
 
@@ -1714,10 +1724,18 @@ Channel::StopSend()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::StopSend()");
+    {
+        // A lock is needed because |_sending| can be accessed or modified by
+        // another thread at the same time.
+        CriticalSectionScoped cs(_callbackCritSect);
+
         if (!_sending)
         {
             return 0;
         }
+        _sending = false;
+    }
+
     // Reset sending SSRC and sequence number and triggers direct transmission
     // of RTCP BYE
     if (_rtpRtcpModule.SetSendingStatus(false) == -1 ||
@@ -1728,7 +1746,6 @@ Channel::StopSend()
             "StartSend() RTP/RTCP failed to stop sending");
     }
 
-    _sending = false;
     return 0;
 }
 

src/voice_engine/main/source/channel.h

@@ -464,52 +464,56 @@ public:
     WebRtc_UWord32 InstanceId() const
     {
         return _instanceId;
-    };
+    }
     WebRtc_Word32 ChannelId() const
     {
         return _channelId;
-    };
+    }
     bool Playing() const
     {
         return _playing;
-    };
+    }
     bool Sending() const
     {
+        // A lock is needed because |_sending| is accessed by both
+        // TransmitMixer::PrepareDemux() and StartSend()/StopSend(), which
+        // are called by different threads.
+        CriticalSectionScoped cs(_callbackCritSect);
         return _sending;
-    };
+    }
     bool Receiving() const
     {
         return _receiving;
-    };
+    }
     bool ExternalTransport() const
     {
         return _externalTransport;
-    };
+    }
     bool OutputIsOnHold() const
     {
         return _outputIsOnHold;
-    };
+    }
     bool InputIsOnHold() const
     {
         return _inputIsOnHold;
-    };
+    }
     RtpRtcp* RtpRtcpModulePtr() const
     {
         return &_rtpRtcpModule;
-    };
+    }
     WebRtc_Word8 OutputEnergyLevel() const
     {
         return _outputAudioLevel.Level();
-    };
+    }
 #ifndef WEBRTC_EXTERNAL_TRANSPORT
     bool SendSocketsInitialized() const
     {
         return _socketTransportModule.SendSocketsInitialized();
-    };
+    }
     bool ReceiveSocketsInitialized() const
     {
         return _socketTransportModule.ReceiveSocketsInitialized();
-    };
+    }
 #endif
     WebRtc_UWord32 Demultiplex(const AudioFrame& audioFrame);
     WebRtc_UWord32 PrepareEncodeAndSend(int mixingFrequency);