Enable cpplint for webrtc/video_engine

Enable cpplint and have it use a whitelist that also checks
in subdirectories.

Move the cpplint check so it runs before the pylint check
since that one always run and increases the time to errors
for cpplint.

Fix all cpplint errors in webrtc/video_engine.

BUG=webrtc:5149
TESTED=Fixed issues reported by:
find webrtc/video_engine -type f -name *.cc -o -name *.h | xargs cpplint.py
followed by 'git cl presubmit'.

R=pbos@chromium.org, phoglund@chromium.org
TBR=pbos@webrtc.org, phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/1481723003 .

Cr-Commit-Position: refs/heads/master@{#10808}
diff --git a/webrtc/video_engine/overuse_frame_detector.h b/webrtc/video_engine/overuse_frame_detector.h
index 0deb5ba..050b14f 100644
--- a/webrtc/video_engine/overuse_frame_detector.h
+++ b/webrtc/video_engine/overuse_frame_detector.h
@@ -50,7 +50,7 @@
   // Method based on encode time of frames.
   bool enable_encode_usage_method;
   int low_encode_usage_threshold_percent;  // Threshold for triggering underuse.
-  int high_encode_usage_threshold_percent; // Threshold for triggering overuse.
+  int high_encode_usage_threshold_percent;  // Threshold for triggering overuse.
   bool enable_extended_processing_usage;  // Include a larger time span (in
                                           // addition to encode time) for
                                           // measuring the processing time of a
@@ -61,9 +61,9 @@
   int min_frame_samples;  // The minimum number of frames required.
   int min_process_count;  // The number of initial process times required before
                           // triggering an overuse/underuse.
-  int high_threshold_consecutive_count; // The number of consecutive checks
-                                        // above the high threshold before
-                                        // triggering an overuse.
+  int high_threshold_consecutive_count;  // The number of consecutive checks
+                                         // above the high threshold before
+                                         // triggering an overuse.
 };
 
 struct CpuOveruseMetrics {
@@ -71,9 +71,9 @@
       : avg_encode_time_ms(-1),
         encode_usage_percent(-1) {}
 
-  int avg_encode_time_ms;   // The average encode time in ms.
-  int encode_usage_percent; // The average encode time divided by the average
-                            // time difference between incoming captured frames.
+  int avg_encode_time_ms;    // Average encode time in ms.
+  int encode_usage_percent;  // Average encode time divided by the average time
+                             // difference between incoming captured frames.
 };
 
 class CpuOveruseMetricsObserver {
diff --git a/webrtc/video_engine/overuse_frame_detector_unittest.cc b/webrtc/video_engine/overuse_frame_detector_unittest.cc
index d502f02..5524ba5 100644
--- a/webrtc/video_engine/overuse_frame_detector_unittest.cc
+++ b/webrtc/video_engine/overuse_frame_detector_unittest.cc
@@ -188,7 +188,7 @@
   ReinitializeOveruseDetector();
   EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(0);
   EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(64);
-  for(size_t i = 0; i < 64; ++i) {
+  for (size_t i = 0; i < 64; ++i) {
     TriggerOveruse(options_.high_threshold_consecutive_count);
   }
 }
diff --git a/webrtc/video_engine/payload_router_unittest.cc b/webrtc/video_engine/payload_router_unittest.cc
index acaa400..11c664b 100644
--- a/webrtc/video_engine/payload_router_unittest.cc
+++ b/webrtc/video_engine/payload_router_unittest.cc
@@ -182,7 +182,7 @@
 
   const uint32_t bitrate_1 = 10000;
   const uint32_t bitrate_2 = 76543;
-  std::vector<uint32_t> bitrates (2, bitrate_1);
+  std::vector<uint32_t> bitrates(2, bitrate_1);
   bitrates[1] = bitrate_2;
   EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
       .Times(1);
diff --git a/webrtc/video_engine/stream_synchronization_unittest.cc b/webrtc/video_engine/stream_synchronization_unittest.cc
index 7136f1e..1c55c2b 100644
--- a/webrtc/video_engine/stream_synchronization_unittest.cc
+++ b/webrtc/video_engine/stream_synchronization_unittest.cc
@@ -8,9 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <algorithm>
 #include <math.h>
 
+#include <algorithm>
+
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/video_engine/stream_synchronization.h"
 
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 2e56056..91c862b 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -11,6 +11,7 @@
 #include "webrtc/video_engine/vie_channel.h"
 
 #include <algorithm>
+#include <map>
 #include <vector>
 
 #include "webrtc/base/checks.h"
@@ -60,7 +61,7 @@
 
 class ViEChannelProtectionCallback : public VCMProtectionCallback {
  public:
-  ViEChannelProtectionCallback(ViEChannel* owner) : owner_(owner) {}
+  explicit ViEChannelProtectionCallback(ViEChannel* owner) : owner_(owner) {}
   ~ViEChannelProtectionCallback() {}
 
 
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index 32f0301..d5030de 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -12,6 +12,8 @@
 #define WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
 
 #include <list>
+#include <map>
+#include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
diff --git a/webrtc/video_engine/vie_receiver.h b/webrtc/video_engine/vie_receiver.h
index d75622f..b0c5749 100644
--- a/webrtc/video_engine/vie_receiver.h
+++ b/webrtc/video_engine/vie_receiver.h
@@ -12,6 +12,7 @@
 #define WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
 
 #include <list>
+#include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/engine_configurations.h"
@@ -84,6 +85,7 @@
   bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
 
   ReceiveStatistics* GetReceiveStatistics() const;
+
  private:
   int InsertRTPPacket(const uint8_t* rtp_packet, size_t rtp_packet_length,
                       const PacketTime& packet_time);
@@ -125,6 +127,6 @@
   int64_t last_packet_log_ms_;
 };
 
-}  // namespace webrt
+}  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
diff --git a/webrtc/video_engine/vie_remb_unittest.cc b/webrtc/video_engine/vie_remb_unittest.cc
index b6fbf29..f319d0d 100644
--- a/webrtc/video_engine/vie_remb_unittest.cc
+++ b/webrtc/video_engine/vie_remb_unittest.cc
@@ -11,11 +11,10 @@
 
 // This file includes unit tests for ViERemb.
 
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
-
 #include <vector>
 
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"