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Change explicit static cast from int to uint16_t to implicit cast of 0u.

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BUG=3663
TESTED=local windows build with VS2013.
R=harryjin@google.com, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7123 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
fbarchard@google.com
2014-09-09 21:37:27 +00:00
parent 9fe11010f7
commit a941970d4a
1 changed files with 5 additions and 5 deletions
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10
webrtc/modules/audio_coding/main/test/RTPFile.cc
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@ -109,7 +109,7 @@ uint16_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
if (packet->payloadSize > 0 && payloadSize >= packet->payloadSize) {
memcpy(payloadData, packet->payloadData, packet->payloadSize);
} else {
return 0;
return 0u;
}
*offset = (packet->timeStamp / (packet->frequency / 1000));
@ -216,7 +216,7 @@ uint16_t RTPFile::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
/* Check if we have reached end of file. */
if ((read_len == 0) && feof(_rtpFile)) {
_rtpEOF = true;
return 0;
return 0u;
}
EXPECT_EQ(1u, fread(&plen, 2, 1, _rtpFile));
EXPECT_EQ(1u, fread(offset, 4, 1, _rtpFile));
@ -232,13 +232,13 @@ uint16_t RTPFile::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
EXPECT_EQ(lengthBytes, plen + 8);
if (plen == 0) {
return static_cast<uint16_t>(0);
return 0u;
}
if (payloadSize < (lengthBytes - 20)) {
return static_cast<uint16_t>(0);
return 0u;
}
if (lengthBytes < 20) {
return static_cast<uint16_t>(0);
return 0u;
}
lengthBytes -= 20;
EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile));