Lint clean video/ and add lint presubmit check.

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <algorithm>
@@ skipping 30 matching lines @@
 #include "webrtc/test/null_transport.h"
 #include "webrtc/test/rtcp_packet_parser.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/test/testsupport/perf_test.h"
 #include "webrtc/video_encoder.h"
 
 namespace webrtc {
 
-static const unsigned long kSilenceTimeoutMs = 2000;
+static const uint32_t kSilenceTimeoutMs = 2000;
 
 class EndToEndTest : public test::CallTest {
  public:
   EndToEndTest() {}
 
   virtual ~EndToEndTest() {
     EXPECT_EQ(nullptr, send_stream_);
     EXPECT_TRUE(receive_streams_.empty());
   }
 
@@ skipping 489 matching lines @@
     std::set<uint32_t> protected_timestamps_ GUARDED_BY(crit_);
   } test;
 
   RunBaseTest(&test, FakeNetworkPipe::Config());
 }
 
 // Flacky on all platforms. See webrtc:4328.
 TEST_F(EndToEndTest, DISABLED_ReceivedFecPacketsNotNacked) {
   class FecNackObserver : public test::EndToEndTest {
    public:
-    explicit FecNackObserver()
+    FecNackObserver()
         : EndToEndTest(kDefaultTimeoutMs),
           state_(kFirstPacket),
           fec_sequence_number_(0),
           has_last_sequence_number_(false),
           last_sequence_number_(0) {}
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock_(&crit_);
       RTPHeader header;
@@ skipping 2468 matching lines @@
       int initial_sender_rtcp;
       int initial_receiver_rtcp;
       {
         rtc::CritScope lock(&test_crit_);
         initial_sender_rtp = sender_rtp_;
         initial_sender_rtcp = sender_rtcp_;
         initial_receiver_rtcp = receiver_rtcp_;
       }
       bool sender_done = false;
       bool receiver_done = false;
-      while(!sender_done || !receiver_done) {
+      while (!sender_done || !receiver_done) {
         packet_event_->Wait(kSilenceTimeoutMs);
         int64_t time_now_ms = clock_->TimeInMilliseconds();
         rtc::CritScope lock(&test_crit_);
         if (sender_down) {
           ASSERT_LE(sender_rtp_ - initial_sender_rtp, kNumAcceptedDowntimeRtp)
               << "RTP sent during sender-side downtime.";
           ASSERT_LE(sender_rtcp_ - initial_sender_rtcp,
                     kNumAcceptedDowntimeRtcp)
               << "RTCP sent during sender-side downtime.";
           if (time_now_ms - initial_time_ms >=
@@ skipping 67 matching lines @@
     }
     SleepMs(10);
   }
 
   Stop();
   DestroyStreams();
 }
 
 TEST_F(EndToEndTest, NewSendStreamsRespectNetworkDown) {
   class UnusedEncoder : public test::FakeEncoder {
-    public:
+   public:
      UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}
      int32_t Encode(const VideoFrame& input_image,
                     const CodecSpecificInfo* codec_specific_info,
                     const std::vector<FrameType>* frame_types) override {
       ADD_FAILURE() << "Unexpected frame encode.";
       return test::FakeEncoder::Encode(
           input_image, codec_specific_info, frame_types);
     }
   };
 
@@ skipping 74 matching lines @@
   EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
 
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyFecConfig(default_receive_config.rtp.fec);
 }
 
 }  // namespace webrtc