Fix some code styles.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22009004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6830 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video/bitrate_estimator_tests.cc b/video/bitrate_estimator_tests.cc
index 1ec95c3..40c1ed6 100644
--- a/video/bitrate_estimator_tests.cc
+++ b/video/bitrate_estimator_tests.cc
@@ -107,7 +107,6 @@
Strings received_log_lines_ GUARDED_BY(crit_sect_);
Strings expected_log_lines_ GUARDED_BY(crit_sect_);
scoped_ptr<EventWrapper> done_;
-
};
Callback callback_;
diff --git a/video/call.cc b/video/call.cc
index bcce6f0..cd29d62 100644
--- a/video/call.cc
+++ b/video/call.cc
@@ -43,7 +43,7 @@
class CpuOveruseObserverProxy : public webrtc::CpuOveruseObserver {
public:
- CpuOveruseObserverProxy(OveruseCallback* overuse_callback)
+ explicit CpuOveruseObserverProxy(OveruseCallback* overuse_callback)
: crit_(CriticalSectionWrapper::CreateCriticalSection()),
overuse_callback_(overuse_callback) {
assert(overuse_callback != NULL);
@@ -327,7 +327,7 @@
return DELIVERY_PACKET_ERROR;
const uint8_t* ptr = &packet[8];
- uint32_t ssrc = ptr[0] << 24 | ptr[1] << 16 | ptr[2] << 8 | ptr[3] ;
+ uint32_t ssrc = ptr[0] << 24 | ptr[1] << 16 | ptr[2] << 8 | ptr[3];
ReadLockScoped read_lock(*receive_lock_);
std::map<uint32_t, VideoReceiveStream*>::iterator it =
diff --git a/video/end_to_end_tests.cc b/video/end_to_end_tests.cc
index 3b5012d..8b64f14 100644
--- a/video/end_to_end_tests.cc
+++ b/video/end_to_end_tests.cc
@@ -1544,7 +1544,7 @@
public:
ObserveRedundantPayloads()
: EndToEndTest(kDefaultTimeoutMs), ssrcs_to_observe_(kNumSsrcs) {
- for(size_t i = 0; i < kNumSsrcs; ++i) {
+ for (size_t i = 0; i < kNumSsrcs; ++i) {
registered_rtx_ssrc_[kSendRtxSsrcs[i]] = true;
}
}
@@ -1617,7 +1617,7 @@
static const uint64_t kMaxTimestampGap = kDefaultTimeoutMs * 90;
class RtpSequenceObserver : public test::RtpRtcpObserver {
public:
- RtpSequenceObserver(bool use_rtx)
+ explicit RtpSequenceObserver(bool use_rtx)
: test::RtpRtcpObserver(kDefaultTimeoutMs),
crit_(CriticalSectionWrapper::CreateCriticalSection()),
ssrcs_to_observe_(kNumSsrcs) {
diff --git a/video/transport_adapter.h b/video/transport_adapter.h
index 79f995b..f7cba1d 100644
--- a/video/transport_adapter.h
+++ b/video/transport_adapter.h
@@ -7,8 +7,8 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_VIDEO_ENGINE_INTERNAL_TRANSPORT_ADAPTER_H_
-#define WEBRTC_VIDEO_ENGINE_INTERNAL_TRANSPORT_ADAPTER_H_
+#ifndef WEBRTC_VIDEO_TRANSPORT_ADAPTER_H_
+#define WEBRTC_VIDEO_TRANSPORT_ADAPTER_H_
#include "webrtc/common_types.h"
#include "webrtc/system_wrappers/interface/atomic32.h"
@@ -36,4 +36,4 @@
} // namespace internal
} // namespace webrtc
-#endif // WEBRTC_VIDEO_ENGINE_INTERNAL_TRANSPORT_ADAPTER_H_
+#endif // WEBRTC_VIDEO_TRANSPORT_ADAPTER_H_
diff --git a/video/video_receive_stream.h b/video/video_receive_stream.h
index 2a3c6df..c45ebac 100644
--- a/video/video_receive_stream.h
+++ b/video/video_receive_stream.h
@@ -40,7 +40,6 @@
class VideoReceiveStream : public webrtc::VideoReceiveStream,
public I420FrameCallback,
public VideoRenderCallback {
-
public:
VideoReceiveStream(webrtc::VideoEngine* video_engine,
const VideoReceiveStream::Config& config,
diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc
index 45e5c69..4e6b2ba 100644
--- a/video/video_send_stream.cc
+++ b/video/video_send_stream.cc
@@ -10,6 +10,7 @@
#include "webrtc/video/video_send_stream.h"
+#include <algorithm>
#include <sstream>
#include <string>
#include <vector>
diff --git a/video/video_send_stream.h b/video/video_send_stream.h
index b0cf0c6..f1f8f7d 100644
--- a/video/video_send_stream.h
+++ b/video/video_send_stream.h
@@ -12,6 +12,7 @@
#define WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_
#include <map>
+#include <vector>
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index 76739a3..b08c405 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -156,11 +156,11 @@
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
}
+
virtual void PerformTest() OVERRIDE {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out while waiting for single RTP packet.";
}
-
} test;
RunBaseTest(&test);
