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Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors.

Browse Source 
BUG=webrtc:5268,webrtc:5273
TESTED=Fixed issues reported by:
find webrtc/audio -type f -name *.cc -o -name *.h | xargs cpplint.py
find webrtc/call -type f -name *.cc -o -name *.h | xargs cpplint.py
followed by 'git cl presubmit'.

R=kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1483323002 .

Cr-Commit-Position: refs/heads/master@{#10853}
This commit is contained in:
Fredrik Solenberg
2015-12-01 11:26:34 +01:00
parent 0de97f1b74
commit ea07373a2e
5 changed files with 9 additions and 1 deletions
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2
PRESUBMIT.py
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@ -16,6 +16,8 @@ import sys
# Directories that will be scanned by cpplint by the presubmit script. # Directories that will be scanned by cpplint by the presubmit script.
CPPLINT_DIRS = [ CPPLINT_DIRS = [
'webrtc/audio',
'webrtc/call',
'webrtc/video_engine', 'webrtc/video_engine',
] ]

2
webrtc/audio/audio_receive_stream_unittest.cc
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@ -8,6 +8,8 @@
* 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.
*/ */
#include <string>
#include "testing/gtest/include/gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio/audio_receive_stream.h" #include "webrtc/audio/audio_receive_stream.h"

3
webrtc/audio/audio_send_stream_unittest.cc
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@ -8,6 +8,9 @@
* 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.
*/ */
#include <string>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio/audio_send_stream.h" #include "webrtc/audio/audio_send_stream.h"

2
webrtc/call/call.cc
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@ -126,7 +126,7 @@ class Call : public webrtc::Call, public PacketReceiver,
Call::Config config_; Call::Config config_;
rtc::ThreadChecker configuration_thread_checker_; rtc::ThreadChecker configuration_thread_checker_;
bool network_enabled_; bool network_enabled_;
rtc::scoped_ptr<RWLockWrapper> receive_crit_; rtc::scoped_ptr<RWLockWrapper> receive_crit_;
// Audio and Video receive streams are owned by the client that creates them. // Audio and Video receive streams are owned by the client that creates them.

1
webrtc/call/rtc_event_log_unittest.cc
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@ -11,6 +11,7 @@
#ifdef ENABLE_RTC_EVENT_LOG #ifdef ENABLE_RTC_EVENT_LOG
#include <string> #include <string>
#include <utility>
#include <vector> #include <vector>
#include "testing/gtest/include/gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"