commit ea07373a2eb46f2732a8b5acef06a9b5078f37f8
author Fredrik Solenberg <solenberg@webrtc.org> Tue Dec 01 11:26:34 2015 +0100
committer Fredrik Solenberg <solenberg@webrtc.org> Tue Dec 01 10:26:46 2015 +0000
tree ebe8d112f0d0972d4c3a313b13a561ba8e0a3954
parent 0de97f1b748d8238fe3a7ad8d7afb2b6cb456a3e

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

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}

PRESUBMIT.py

diff --git a/PRESUBMIT.py b/PRESUBMIT.py
index 5258549..06a1c4a 100755
--- a/PRESUBMIT.py
+++ b/PRESUBMIT.py
@@ -16,6 +16,8 @@
 
 # Directories that will be scanned by cpplint by the presubmit script.
 CPPLINT_DIRS = [
+  'webrtc/audio',
+  'webrtc/call',
   'webrtc/video_engine',
 ]
 

webrtc/audio/audio_receive_stream_unittest.cc

diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 40c2d64..bfd8dca 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <string>
+
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/audio/audio_receive_stream.h"

webrtc/audio/audio_send_stream_unittest.cc

diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index eed97c7..8dc6da7 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -8,6 +8,9 @@
  *  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 "webrtc/audio/audio_send_stream.h"

webrtc/call/call.cc

diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 4d758d9..441de87 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -126,7 +126,7 @@
   Call::Config config_;
   rtc::ThreadChecker configuration_thread_checker_;
 
- bool network_enabled_;
+  bool network_enabled_;
 
   rtc::scoped_ptr<RWLockWrapper> receive_crit_;
   // Audio and Video receive streams are owned by the client that creates them.

webrtc/call/rtc_event_log_unittest.cc

diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc
index 9dc7bec..eb3c148 100644
--- a/webrtc/call/rtc_event_log_unittest.cc
+++ b/webrtc/call/rtc_event_log_unittest.cc
@@ -11,6 +11,7 @@
 #ifdef ENABLE_RTC_EVENT_LOG
 
 #include <string>
+#include <utility>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"