Fixing lint errors in NetEq4
Just taking care of a few old lint errors.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5359 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
index 965f75f..bf5ca7b 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
@@ -999,7 +999,7 @@
// Even if there is RTP packet in NetEq's buffer, the first frame pulled
// from NetEq starts with few zero samples. Here we measure this delay.
if (n == 0) {
- while(decoded[delay_samples] == 0) delay_samples++;
+ while (decoded[delay_samples] == 0) delay_samples++;
}
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
@@ -1182,7 +1182,6 @@
// Expect delay (in samples) to be less than 2 packets.
EXPECT_LE(timestamp - neteq_->PlayoutTimestamp(),
static_cast<uint32_t>(kSamples * 2));
-
}
// Make sure we have actually tested wrap-around.
ASSERT_EQ(expect_seq_no_wrap, seq_no_wrapped);
@@ -1216,4 +1215,4 @@
WrapTest(0xFFFF - 10, 0xFFFFFFFF - 5000, drop_seq_numbers, true, true);
}
-} // namespace
+} // namespace webrtc