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/audio_multi_vector.cc b/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc
index baa912c..b49f8b0 100644
--- a/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc
+++ b/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc
@@ -74,8 +74,7 @@
return;
}
size_t length_per_channel = length / num_channels_;
- int16_t* temp_array =
- new int16_t[length_per_channel]; // Intermediate storage.
+ int16_t* temp_array = new int16_t[length_per_channel]; // Temporary storage.
for (size_t channel = 0; channel < num_channels_; ++channel) {
// Copy elements to |temp_array|.
// Set |source_ptr| to first element of this channel.
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
index fb27af2..cd69fc9 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
@@ -143,12 +143,12 @@
int error = InsertPacketInternal(
rtp_header, kSyncPayload, sizeof(kSyncPayload), receive_timestamp, true);
- if (error != 0) {
- LOG_FERR1(LS_WARNING, InsertPacketInternal, error);
- error_code_ = error;
- return kFail;
- }
- return kOK;
+ if (error != 0) {
+ LOG_FERR1(LS_WARNING, InsertPacketInternal, error);
+ error_code_ = error;
+ return kFail;
+ }
+ return kOK;
}
int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
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
diff --git a/webrtc/modules/audio_coding/neteq4/normal.h b/webrtc/modules/audio_coding/neteq4/normal.h
index fa14685..df28319 100644
--- a/webrtc/modules/audio_coding/neteq4/normal.h
+++ b/webrtc/modules/audio_coding/neteq4/normal.h
@@ -65,4 +65,4 @@
};
} // namespace webrtc
-#endif // SRC_MODULES_AUDIO_CODING_NETEQ4_NORMAL_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_NORMAL_H_
diff --git a/webrtc/modules/audio_coding/neteq4/random_vector.cc b/webrtc/modules/audio_coding/neteq4/random_vector.cc
index 823909f..e7a5a1d 100644
--- a/webrtc/modules/audio_coding/neteq4/random_vector.cc
+++ b/webrtc/modules/audio_coding/neteq4/random_vector.cc
@@ -54,4 +54,4 @@
seed_increment_+= increase_by;
seed_increment_ &= kRandomTableSize - 1;
}
-}
+} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq4/rtcp.cc b/webrtc/modules/audio_coding/neteq4/rtcp.cc
index f9dcf44..bc178fc 100644
--- a/webrtc/modules/audio_coding/neteq4/rtcp.cc
+++ b/webrtc/modules/audio_coding/neteq4/rtcp.cc
@@ -10,9 +10,10 @@
#include "webrtc/modules/audio_coding/neteq4/rtcp.h"
-#include <algorithm>
#include <string.h>
+#include <algorithm>
+
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/interface/module_common_types.h"