Fixe issues reported by Coverity for modules/utility.
BUG=From Coverity TEST=N/A Review URL: https://webrtc-codereview.appspot.com/389011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1695 4adac7df-926f-26a2-2b94-8c16560cd09d
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		| @@ -64,10 +64,12 @@ FilePlayerImpl::FilePlayerImpl(const WebRtc_UWord32 instanceID, | ||||
|       _fileFormat(fileFormat), | ||||
|       _fileModule(*MediaFile::CreateMediaFile(instanceID)), | ||||
|       _decodedLengthInMS(0), | ||||
|       _decodedAudioBuffer(), | ||||
|       _audioDecoder(instanceID), | ||||
|       _codec(), | ||||
|       _numberOf10MsPerFrame(0), | ||||
|       _numberOf10MsInDecoder(0), | ||||
|       _resampler(), | ||||
|       _scaling(1.0) | ||||
| { | ||||
|     _codec.plfreq = 0; | ||||
| @@ -466,11 +468,14 @@ VideoFilePlayerImpl::VideoFilePlayerImpl(WebRtc_UWord32 instanceID, | ||||
|                                          FileFormats fileFormat) | ||||
|     : FilePlayerImpl(instanceID,fileFormat), | ||||
|       _videoDecoder(*new VideoCoder(instanceID)), | ||||
|       video_codec_info_(), | ||||
|       _decodedVideoFrames(0), | ||||
|       _encodedData(*new EncodedVideoData()), | ||||
|       _frameScaler(*new FrameScaler()), | ||||
|       _critSec(CriticalSectionWrapper::CreateCriticalSection()), | ||||
|       _startTime(), | ||||
|       _accumulatedRenderTimeMs(0), | ||||
|       _frameLengthMS(0), | ||||
|       _numberOfFramesRead(0), | ||||
|       _videoOnly(false) | ||||
| { | ||||
|   | ||||
| @@ -65,8 +65,12 @@ FileRecorderImpl::FileRecorderImpl(WebRtc_UWord32 instanceID, | ||||
|     : _instanceID(instanceID), | ||||
|       _fileFormat(fileFormat), | ||||
|       _moduleFile(MediaFile::CreateMediaFile(_instanceID)), | ||||
|       _stream(NULL), | ||||
|       codec_info_(), | ||||
|       _amrFormat(AMRFileStorage), | ||||
|       _audioEncoder(instanceID) | ||||
|       _audioBuffer(), | ||||
|       _audioEncoder(instanceID), | ||||
|       _audioResampler() | ||||
| { | ||||
| } | ||||
|  | ||||
| @@ -353,7 +357,8 @@ class AudioFrameFileInfo | ||||
|                      const WebRtc_UWord16 audioSize, | ||||
|                      const WebRtc_UWord16 audioMS, | ||||
|                      const TickTime& playoutTS) | ||||
|            : _audioSize(audioSize), _audioMS(audioMS) ,_playoutTS(playoutTS) | ||||
|            : _audioData(), _audioSize(audioSize), _audioMS(audioMS), | ||||
|              _playoutTS(playoutTS) | ||||
|        { | ||||
|            if(audioSize > MAX_AUDIO_BUFFER_IN_BYTES) | ||||
|            { | ||||
|   | ||||
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	 henrike@webrtc.org
					henrike@webrtc.org