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Fix some code styles.

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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@6830 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2014-08-05 23:35:43 +00:00
parent 624a504f5b
commit dde16f19e3
8 changed files with 10 additions and 10 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
webrtc/video/bitrate_estimator_tests.cc
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@ -107,7 +107,6 @@ class TraceObserver {
Strings received_log_lines_ GUARDED_BY(crit_sect_);
Strings expected_log_lines_ GUARDED_BY(crit_sect_);
scoped_ptr<EventWrapper> done_;
};
Callback callback_;

2
webrtc/video/call.cc
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@ -43,7 +43,7 @@ namespace internal {
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);

2
webrtc/video/end_to_end_tests.cc
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@ -1617,7 +1617,7 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
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) {

6
webrtc/video/transport_adapter.h
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@ -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 @@ class TransportAdapter : public webrtc::Transport {
} // namespace internal
} // namespace webrtc
#endif // WEBRTC_VIDEO_ENGINE_INTERNAL_TRANSPORT_ADAPTER_H_
#endif // WEBRTC_VIDEO_TRANSPORT_ADAPTER_H_

1
webrtc/video/video_receive_stream.h
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@ -40,7 +40,6 @@ namespace internal {
class VideoReceiveStream : public webrtc::VideoReceiveStream,
public I420FrameCallback,
public VideoRenderCallback {
public:
VideoReceiveStream(webrtc::VideoEngine* video_engine,
const VideoReceiveStream::Config& config,

1
webrtc/video/video_send_stream.cc
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@ -10,6 +10,7 @@
#include "webrtc/video/video_send_stream.h"
#include <algorithm>
#include <sstream>
#include <string>
#include <vector>

1
webrtc/video/video_send_stream.h
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@ -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"

2
webrtc/video/video_send_stream_tests.cc
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@ -156,11 +156,11 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
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);