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Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable.

Browse Source 
BUG=2944
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5579 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
turaj@webrtc.org 2014-02-19 20:31:17 +00:00
parent 97e7a640d8
commit c2d69d3229
2 changed files with 7 additions and 0 deletions
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5
webrtc/modules/audio_coding/main/test/Channel.cc
View File

@ -120,6 +120,7 @@ int32_t Channel::SendData(const FrameType frameType, const uint8_t payloadType,
return status; return status;
} }
// TODO(turajs): rewite this method.
void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize) { void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize) {
int n; int n;
if ((rtpInfo.header.payloadType != _lastPayloadType) if ((rtpInfo.header.payloadType != _lastPayloadType)
@ -188,6 +189,8 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize) {
currentPayloadStr->lastPayloadLenByte = payloadSize; currentPayloadStr->lastPayloadLenByte = payloadSize;
currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp; currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
currentPayloadStr->payloadType = rtpInfo.header.payloadType; currentPayloadStr->payloadType = rtpInfo.header.payloadType;
memset(currentPayloadStr->frameSizeStats, 0,
sizeof(ACMTestPayloadStats::frameSizeStats));
} }
} else { } else {
n = 0; n = 0;
@ -199,6 +202,8 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize) {
_payloadStats[n].lastPayloadLenByte = payloadSize; _payloadStats[n].lastPayloadLenByte = payloadSize;
_payloadStats[n].lastTimestamp = rtpInfo.header.timestamp; _payloadStats[n].lastTimestamp = rtpInfo.header.timestamp;
_payloadStats[n].payloadType = rtpInfo.header.payloadType; _payloadStats[n].payloadType = rtpInfo.header.payloadType;
memset(_payloadStats[n].frameSizeStats, 0,
sizeof(ACMTestPayloadStats::frameSizeStats));
} }
} }

2
webrtc/modules/audio_coding/main/test/Channel.h
View File

@ -24,6 +24,7 @@ class CriticalSectionWrapper;
#define MAX_NUM_PAYLOADS 50 #define MAX_NUM_PAYLOADS 50
#define MAX_NUM_FRAMESIZES 6 #define MAX_NUM_FRAMESIZES 6
// TODO(turajs): Write constructor for this structure.
struct ACMTestFrameSizeStats { struct ACMTestFrameSizeStats {
uint16_t frameSizeSample; uint16_t frameSizeSample;
int16_t maxPayloadLen; int16_t maxPayloadLen;
@ -34,6 +35,7 @@ struct ACMTestFrameSizeStats {
double usageLenSec; double usageLenSec;
}; };
// TODO(turajs): Write constructor for this structure.
struct ACMTestPayloadStats { struct ACMTestPayloadStats {
bool newPacket; bool newPacket;
int16_t payloadType; int16_t payloadType;