Change explicit static cast from int to uint16_t to implicit cast of 0u.
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
diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.cc b/webrtc/modules/audio_coding/main/test/RTPFile.cc
index 789dc1d..4cef592 100644
--- a/webrtc/modules/audio_coding/main/test/RTPFile.cc
+++ b/webrtc/modules/audio_coding/main/test/RTPFile.cc
@@ -109,7 +109,7 @@
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 @@
/* 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 @@
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));