generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Fix some code styles.

Browse Source 
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
View File

@ -107,7 +107,6 @@ class TraceObserver {
Strings received_log_lines_ GUARDED_BY(crit_sect_); Strings received_log_lines_ GUARDED_BY(crit_sect_);
Strings expected_log_lines_ GUARDED_BY(crit_sect_); Strings expected_log_lines_ GUARDED_BY(crit_sect_);
scoped_ptr<EventWrapper> done_; scoped_ptr<EventWrapper> done_;
}; };
Callback callback_; Callback callback_;

2
webrtc/video/call.cc
View File

@ -43,7 +43,7 @@ namespace internal {
class CpuOveruseObserverProxy : public webrtc::CpuOveruseObserver { class CpuOveruseObserverProxy : public webrtc::CpuOveruseObserver {
public: public:
CpuOveruseObserverProxy(OveruseCallback* overuse_callback) explicit CpuOveruseObserverProxy(OveruseCallback* overuse_callback)
: crit_(CriticalSectionWrapper::CreateCriticalSection()), : crit_(CriticalSectionWrapper::CreateCriticalSection()),
overuse_callback_(overuse_callback) { overuse_callback_(overuse_callback) {
assert(overuse_callback != NULL); assert(overuse_callback != NULL);

2
webrtc/video/end_to_end_tests.cc
View File

@ -1617,7 +1617,7 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
static const uint64_t kMaxTimestampGap = kDefaultTimeoutMs * 90; static const uint64_t kMaxTimestampGap = kDefaultTimeoutMs * 90;
class RtpSequenceObserver : public test::RtpRtcpObserver { class RtpSequenceObserver : public test::RtpRtcpObserver {
public: public:
RtpSequenceObserver(bool use_rtx) explicit RtpSequenceObserver(bool use_rtx)
: test::RtpRtcpObserver(kDefaultTimeoutMs), : test::RtpRtcpObserver(kDefaultTimeoutMs),
crit_(CriticalSectionWrapper::CreateCriticalSection()), crit_(CriticalSectionWrapper::CreateCriticalSection()),
ssrcs_to_observe_(kNumSsrcs) { ssrcs_to_observe_(kNumSsrcs) {

6
webrtc/video/transport_adapter.h
View File

@ -7,8 +7,8 @@
* in the file PATENTS. All contributing project authors may * in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_VIDEO_ENGINE_INTERNAL_TRANSPORT_ADAPTER_H_ #ifndef WEBRTC_VIDEO_TRANSPORT_ADAPTER_H_
#define WEBRTC_VIDEO_ENGINE_INTERNAL_TRANSPORT_ADAPTER_H_ #define WEBRTC_VIDEO_TRANSPORT_ADAPTER_H_
#include "webrtc/common_types.h" #include "webrtc/common_types.h"
#include "webrtc/system_wrappers/interface/atomic32.h" #include "webrtc/system_wrappers/interface/atomic32.h"
@ -36,4 +36,4 @@ class TransportAdapter : public webrtc::Transport {
} // namespace internal } // namespace internal
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_VIDEO_ENGINE_INTERNAL_TRANSPORT_ADAPTER_H_ #endif // WEBRTC_VIDEO_TRANSPORT_ADAPTER_H_

1
webrtc/video/video_receive_stream.h
View File

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

1
webrtc/video/video_send_stream.cc
View File

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

1
webrtc/video/video_send_stream.h
View File

@ -12,6 +12,7 @@
#define WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_ #define WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_
#include <map> #include <map>
#include <vector>
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"

2
webrtc/video/video_send_stream_tests.cc
View File

@ -156,11 +156,11 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
send_config->rtp.extensions.push_back( send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId)); RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
} }
virtual void PerformTest() OVERRIDE { virtual void PerformTest() OVERRIDE {
EXPECT_EQ(kEventSignaled, Wait()) EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out while waiting for single RTP packet."; << "Timed out while waiting for single RTP packet.";
} }
} test; } test;
RunBaseTest(&test); RunBaseTest(&test);