Simple CL to fix lint errors in webrtc/modules/remote_bitrate_estimator. Added the lint check for the folder to the presubmit script.

BUG=webrtc:5310

Review URL: https://codereview.webrtc.org/1520513003

Cr-Commit-Position: refs/heads/master@{#11021}

PRESUBMIT.py

@@ -20,6 +20,7 @@ CPPLINT_DIRS = [
   'webrtc/call',
   'webrtc/common_video',
   'webrtc/examples',
+  'webrtc/modules/remote_bitrate_estimator',
   'webrtc/modules/video_processing',
   'webrtc/sound',
   'webrtc/tools',

webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc

@@ -88,8 +88,7 @@ uint32_t AimdRateControl::LatestEstimate() const {
 
 uint32_t AimdRateControl::UpdateBandwidthEstimate(int64_t now_ms) {
   current_bitrate_bps_ = ChangeBitrate(current_bitrate_bps_,
-                                       current_input_._incomingBitRate,
+                                       current_input_.incoming_bitrate, now_ms);
-                                       now_ms);
   if (now_ms - time_of_last_log_ > kLogIntervalMs) {
     time_of_last_log_ = now_ms;
   }
@@ -109,21 +108,21 @@ void AimdRateControl::Update(const RateControlInput* input, int64_t now_ms) {
     const int64_t kInitializationTimeMs = 5000;
     RTC_DCHECK_LE(kBitrateWindowMs, kInitializationTimeMs);
     if (time_first_incoming_estimate_ < 0) {
-      if (input->_incomingBitRate > 0) {
+      if (input->incoming_bitrate > 0) {
         time_first_incoming_estimate_ = now_ms;
       }
     } else if (now_ms - time_first_incoming_estimate_ > kInitializationTimeMs &&
-               input->_incomingBitRate > 0) {
+               input->incoming_bitrate > 0) {
-      current_bitrate_bps_ = input->_incomingBitRate;
+      current_bitrate_bps_ = input->incoming_bitrate;
       bitrate_is_initialized_ = true;
     }
   }
 
-  if (updated_ && current_input_._bwState == kBwOverusing) {
+  if (updated_ && current_input_.bw_state == kBwOverusing) {
     // Only update delay factor and incoming bit rate. We always want to react
     // on an over-use.
-    current_input_._noiseVar = input->_noiseVar;
+    current_input_.noise_var = input->noise_var;
-    current_input_._incomingBitRate = input->_incomingBitRate;
+    current_input_.incoming_bitrate = input->incoming_bitrate;
   } else {
     updated_ = true;
     current_input_ = *input;
@@ -145,7 +144,7 @@ uint32_t AimdRateControl::ChangeBitrate(uint32_t current_bitrate_bps,
   // An over-use should always trigger us to reduce the bitrate, even though
   // we have not yet established our first estimate. By acting on the over-use,
   // we will end up with a valid estimate.
-  if (!bitrate_is_initialized_ && current_input_._bwState != kBwOverusing)
+  if (!bitrate_is_initialized_ && current_input_.bw_state != kBwOverusing)
     return current_bitrate_bps_;
   updated_ = false;
   ChangeState(current_input_, now_ms);
@@ -284,7 +283,7 @@ void AimdRateControl::UpdateMaxBitRateEstimate(float incoming_bitrate_kbps) {
 
 void AimdRateControl::ChangeState(const RateControlInput& input,
                                   int64_t now_ms) {
-  switch (current_input_._bwState) {
+  switch (current_input_.bw_state) {
     case kBwNormal:
       if (rate_control_state_ == kRcHold) {
         time_last_bitrate_change_ = now_ms;

webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc

@@ -74,7 +74,6 @@ TEST_P(BweSimulation, Verizon4gDownlinkTest) {
 }
 
 TEST_P(BweSimulation, Choke1000kbps500kbps1000kbpsBiDirectional) {
-
   const int kFlowIds[] = {0, 1};
   const size_t kNumFlows = sizeof(kFlowIds) / sizeof(kFlowIds[0]);
 
@@ -106,7 +105,6 @@ TEST_P(BweSimulation, Choke1000kbps500kbps1000kbpsBiDirectional) {
 }
 
 TEST_P(BweSimulation, Choke1000kbps500kbps1000kbps) {
-
   AdaptiveVideoSource source(0, 30, 300, 0, 0);
   VideoSender sender(&uplink_, &source, GetParam());
   ChokeFilter choke(&uplink_, 0);
@@ -243,7 +241,7 @@ TEST_P(BweSimulation, PacerGoogleWifiTrace3Mbps) {
 }
 
 TEST_P(BweSimulation, SelfFairnessTest) {
-  srand(Clock::GetRealTimeClock()->TimeInMicroseconds());
+  Random prng(Clock::GetRealTimeClock()->TimeInMicroseconds());
   const int kAllFlowIds[] = {0, 1, 2, 3};
   const size_t kNumFlows = sizeof(kAllFlowIds) / sizeof(kAllFlowIds[0]);
   rtc::scoped_ptr<VideoSource> sources[kNumFlows];
@@ -252,7 +250,7 @@ TEST_P(BweSimulation, SelfFairnessTest) {
     // Streams started 20 seconds apart to give them different advantage when
     // competing for the bandwidth.
     sources[i].reset(new AdaptiveVideoSource(kAllFlowIds[i], 30, 300, 0,
-                                             i * (rand() % 40000)));
+                                             i * prng.Rand(39999)));
     senders[i].reset(new VideoSender(&uplink_, sources[i].get(), GetParam()));
   }
 

webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
 
 #include "webrtc/typedefs.h"
 
@@ -20,41 +20,28 @@ namespace webrtc {
 
 static const int64_t kBitrateWindowMs = 1000;
 
-enum BandwidthUsage
+enum BandwidthUsage {
-{
   kBwNormal = 0,
   kBwUnderusing = 1,
   kBwOverusing = 2,
 };
 
-enum RateControlState
+enum RateControlState { kRcHold, kRcIncrease, kRcDecrease };
-{
-    kRcHold,
-    kRcIncrease,
-    kRcDecrease
-};
 
-enum RateControlRegion
+enum RateControlRegion { kRcNearMax, kRcAboveMax, kRcMaxUnknown };
-{
-    kRcNearMax,
-    kRcAboveMax,
-    kRcMaxUnknown
-};
 
-class RateControlInput
+struct RateControlInput {
-{
+  RateControlInput(BandwidthUsage bw_state,
-public:
+                   uint32_t incoming_bitrate,
-    RateControlInput(BandwidthUsage bwState,
+                   double noise_var)
-                     uint32_t incomingBitRate,
+      : bw_state(bw_state),
-                     double noiseVar)
+        incoming_bitrate(incoming_bitrate),
-        : _bwState(bwState),
+        noise_var(noise_var) {}
-          _incomingBitRate(incomingBitRate),
-          _noiseVar(noiseVar) {}
 
-    BandwidthUsage  _bwState;
+  BandwidthUsage bw_state;
-    uint32_t      _incomingBitRate;
+  uint32_t incoming_bitrate;
-    double              _noiseVar;
+  double noise_var;
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_

webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_OBSERVER_H_
-#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_OBSERVER_H_
 
 #include <vector>
 
@@ -26,4 +26,4 @@ class MockRemoteBitrateObserver : public RemoteBitrateObserver {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_OBSERVER_H_

webrtc/modules/remote_bitrate_estimator/include/send_time_history.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_TIME_HISTORY_H_
+#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_SEND_TIME_HISTORY_H_
-#define WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_TIME_HISTORY_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_SEND_TIME_HISTORY_H_
 
 #include <map>
 
@@ -45,4 +45,4 @@ class SendTimeHistory {
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_TIME_HISTORY_H_
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_SEND_TIME_HISTORY_H_

webrtc/modules/remote_bitrate_estimator/inter_arrival.cc

@@ -71,8 +71,7 @@ bool InterArrival::ComputeDeltas(uint32_t timestamp,
     current_timestamp_group_.first_timestamp = timestamp;
     current_timestamp_group_.timestamp = timestamp;
     current_timestamp_group_.size = 0;
-  }
+  } else {
-  else {
     current_timestamp_group_.timestamp = LatestTimestamp(
         current_timestamp_group_.timestamp, timestamp);
   }

webrtc/modules/remote_bitrate_estimator/overuse_detector.cc

@@ -10,11 +10,13 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/overuse_detector.h"
 
-#include <algorithm>
-#include <sstream>
 #include <math.h>
 #include <stdlib.h>
 
+#include <algorithm>
+#include <sstream>
+#include <string>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/common.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"

webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc

@@ -9,6 +9,8 @@
  */
 
 #include <math.h>
+
+#include <algorithm>
 #include <cmath>
 #include <cstdlib>
 

webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc

@@ -10,12 +10,13 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/overuse_estimator.h"
 
-#include <algorithm>
 #include <assert.h>
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>
 
+#include <algorithm>
+
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
 

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc

@@ -12,6 +12,8 @@
 
 #include <math.h>
 
+#include <algorithm>
+
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc

@@ -17,7 +17,6 @@ namespace webrtc {
 class RemoteBitrateEstimatorAbsSendTimeTest :
     public RemoteBitrateEstimatorTest {
  public:
-
   RemoteBitrateEstimatorAbsSendTimeTest() {}
   virtual void SetUp() {
     bitrate_estimator_.reset(new RemoteBitrateEstimatorAbsSendTime(

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc

@@ -10,6 +10,8 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
 
+#include <utility>
+
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
@@ -32,7 +34,8 @@ struct RemoteBitrateEstimatorSingleStream::Detector {
                     const OverUseDetectorOptions& options,
                     bool enable_burst_grouping)
       : last_packet_time_ms(last_packet_time_ms),
-          inter_arrival(90 * kTimestampGroupLengthMs, kTimestampToMs,
+        inter_arrival(90 * kTimestampGroupLengthMs,
+                      kTimestampToMs,
                       enable_burst_grouping),
         estimator(options),
         detector(options) {}

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc

@@ -17,7 +17,6 @@ namespace webrtc {
 class RemoteBitrateEstimatorSingleTest :
     public RemoteBitrateEstimatorTest {
  public:
-
   RemoteBitrateEstimatorSingleTest() {}
   virtual void SetUp() {
     bitrate_estimator_.reset(new RemoteBitrateEstimatorSingleStream(

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc

@@ -10,6 +10,7 @@
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h"
 
 #include <algorithm>
+#include <limits>
 #include <utility>
 
 namespace webrtc {
@@ -383,10 +384,10 @@ void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper(
                                         2 * kFrameIntervalAbsSendTime);
     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
                    absolute_send_time, true);
-    IncomingPacket(
+    IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(),
-        kDefaultSsrc, 1000, clock_.TimeInMilliseconds(),
                    timestamp - 90 * kFrameIntervalMs,
-        AddAbsSendTime(absolute_send_time, -int(kFrameIntervalAbsSendTime)),
+                   AddAbsSendTime(absolute_send_time,
+                                  -static_cast<int>(kFrameIntervalAbsSendTime)),
                    true);
   }
   bitrate_estimator_->Process();
@@ -520,8 +521,8 @@ void RemoteBitrateEstimatorTest::TestTimestampGroupingTestHelper() {
   uint32_t timestamp = 0;
   // Initialize absolute_send_time (24 bits) so that it will definitely wrap
   // during the test.
-  uint32_t absolute_send_time =
+  uint32_t absolute_send_time = AddAbsSendTime(
-      AddAbsSendTime((1 << 24), -int(50 * kFrameIntervalAbsSendTime));
+      (1 << 24), -static_cast<int>(50 * kFrameIntervalAbsSendTime));
   // Initial set of frames to increase the bitrate. 6 seconds to have enough
   // time for the first estimate to be generated and for Process() to be called.
   for (int i = 0; i <= 6 * kFramerate; ++i) {
@@ -556,8 +557,10 @@ void RemoteBitrateEstimatorTest::TestTimestampGroupingTestHelper() {
     // Increase time until next batch to simulate over-use.
     clock_.AdvanceTimeMilliseconds(10);
     timestamp += 90 * kFrameIntervalMs - kTimestampGroupLength;
-    absolute_send_time = AddAbsSendTime(absolute_send_time, AddAbsSendTime(
+    absolute_send_time = AddAbsSendTime(
-        kFrameIntervalAbsSendTime, -int(kTimestampGroupLengthAbsSendTime)));
+        absolute_send_time,
+        AddAbsSendTime(kFrameIntervalAbsSendTime,
+                       -static_cast<int>(kTimestampGroupLengthAbsSendTime)));
     bitrate_estimator_->Process();
   }
   EXPECT_TRUE(bitrate_observer_->updated());

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h

@@ -14,6 +14,7 @@
 #include <list>
 #include <map>
 #include <utility>
+#include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/constructormagic.h"

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc

@@ -13,8 +13,10 @@
 #include <unistd.h>
 #endif
 
+#include <algorithm>
 #include <sstream>
 
+#include "webrtc/base/random.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h"
@@ -242,18 +244,20 @@ class BweFeedbackTest
     : public BweTest,
       public ::testing::TestWithParam<BandwidthEstimatorType> {
  public:
-  BweFeedbackTest() : BweTest() {}
+#ifdef WEBRTC_WIN
+  BweFeedbackTest()
+      : BweTest(), random_(Clock::GetRealTimeClock()->TimeInMicroseconds()) {}
+#else
+  BweFeedbackTest()
+      : BweTest(),
+        // Multiply the time by a random-ish odd number derived from the PID.
+        random_((getpid() | 1) *
+                Clock::GetRealTimeClock()->TimeInMicroseconds()) {}
+#endif
   virtual ~BweFeedbackTest() {}
 
  protected:
-  void SetUp() override {
+  Random random_;
-    unsigned int seed = Clock::GetRealTimeClock()->TimeInMicroseconds();
-#ifndef WEBRTC_WIN
-    seed *= getpid();
-#endif
-    srand(seed);
-    BweTest::SetUp();
-  }
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(BweFeedbackTest);
@@ -356,7 +360,7 @@ TEST_P(BweFeedbackTest, PacedSelfFairness50msTest) {
   const int kNumRmcatFlows = 4;
   int64_t offset_ms[kNumRmcatFlows];
   for (int i = 0; i < kNumRmcatFlows; ++i) {
-    offset_ms[i] = std::max(0, 5000 * i + rand() % 2001 - 1000);
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
   }
 
   RunFairnessTest(GetParam(), kNumRmcatFlows, 0, 300, 3000, 50, kRttMs,
@@ -370,7 +374,7 @@ TEST_P(BweFeedbackTest, PacedSelfFairness500msTest) {
   const int kNumRmcatFlows = 4;
   int64_t offset_ms[kNumRmcatFlows];
   for (int i = 0; i < kNumRmcatFlows; ++i) {
-    offset_ms[i] = std::max(0, 5000 * i + rand() % 2001 - 1000);
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
   }
 
   RunFairnessTest(GetParam(), kNumRmcatFlows, 0, 300, 3000, 500, kRttMs,
@@ -384,7 +388,7 @@ TEST_P(BweFeedbackTest, PacedSelfFairness1000msTest) {
   const int kNumRmcatFlows = 4;
   int64_t offset_ms[kNumRmcatFlows];
   for (int i = 0; i < kNumRmcatFlows; ++i) {
-    offset_ms[i] = std::max(0, 5000 * i + rand() % 2001 - 1000);
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
   }
 
   RunFairnessTest(GetParam(), kNumRmcatFlows, 0, 300, 3000, 1000, kRttMs,
@@ -397,7 +401,7 @@ TEST_P(BweFeedbackTest, TcpFairness50msTest) {
 
   int64_t offset_ms[2];  // One TCP, one RMCAT flow.
   for (int i = 0; i < 2; ++i) {
-    offset_ms[i] = std::max(0, 5000 * i + rand() % 2001 - 1000);
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
   }
 
   RunFairnessTest(GetParam(), 1, 1, 300, 2000, 50, kRttMs, kMaxJitterMs,
@@ -410,7 +414,7 @@ TEST_P(BweFeedbackTest, TcpFairness500msTest) {
 
   int64_t offset_ms[2];  // One TCP, one RMCAT flow.
   for (int i = 0; i < 2; ++i) {
-    offset_ms[i] = std::max(0, 5000 * i + rand() % 2001 - 1000);
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
   }
 
   RunFairnessTest(GetParam(), 1, 1, 300, 2000, 500, kRttMs, kMaxJitterMs,
@@ -423,7 +427,7 @@ TEST_P(BweFeedbackTest, TcpFairness1000msTest) {
 
   int64_t offset_ms[2];  // One TCP, one RMCAT flow.
   for (int i = 0; i < 2; ++i) {
-    offset_ms[i] = std::max(0, 5000 * i + rand() % 2001 - 1000);
+    offset_ms[i] = std::max(0, 5000 * i + random_.Rand(-1000, 1000));
   }
 
   RunFairnessTest(GetParam(), 1, 1, 300, 2000, 1000, kRttMs, kMaxJitterMs,

webrtc/modules/remote_bitrate_estimator/test/bwe.h

@@ -11,7 +11,10 @@
 #ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BWE_H_
 #define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BWE_H_
 
+#include <list>
+#include <map>
 #include <sstream>
+#include <string>
 
 #include "webrtc/test/testsupport/gtest_prod_util.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet.h"

webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc

@@ -680,6 +680,7 @@ VideoSource::VideoSource(int flow_id,
       frame_period_ms_(1000.0 / fps),
       bits_per_second_(1000 * kbps),
       frame_size_bytes_(bits_per_second_ / 8 / fps),
+      random_(0x12345678),
       flow_id_(flow_id),
       next_frame_ms_(first_frame_offset_ms),
       next_frame_rand_ms_(0),
@@ -713,9 +714,7 @@ void VideoSource::RunFor(int64_t time_ms, Packets* in_out) {
     const int64_t kRandAmplitude = 2;
     // A variance picked uniformly from {-1, 0, 1} ms is added to the frame
     // timestamp.
-    next_frame_rand_ms_ =
+    next_frame_rand_ms_ = kRandAmplitude * (random_.Rand<float>() - 0.5);
-        kRandAmplitude * static_cast<float>(rand()) / RAND_MAX -
-        kRandAmplitude / 2;
 
     // Ensure frame will not have a negative timestamp.
     int64_t next_frame_ms =

webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h

@@ -17,8 +17,10 @@
 #include <algorithm>
 #include <list>
 #include <numeric>
+#include <set>
 #include <sstream>
 #include <string>
+#include <utility>
 #include <vector>
 
 #include "webrtc/base/common.h"
@@ -44,7 +46,7 @@ class DelayCapHelper;
 
 class RateCounter {
  public:
-  RateCounter(int64_t window_size_ms)
+  explicit RateCounter(int64_t window_size_ms)
       : window_size_us_(1000 * window_size_ms),
         recently_received_packets_(0),
         recently_received_bytes_(0),
@@ -415,6 +417,7 @@ class VideoSource {
   uint32_t frame_size_bytes_;
 
  private:
+  Random random_;
   const int flow_id_;
   int64_t next_frame_ms_;
   int64_t next_frame_rand_ms_;

webrtc/modules/remote_bitrate_estimator/test/metric_recorder.cc

@@ -10,10 +10,10 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
 
-#include "webrtc/modules/remote_bitrate_estimator/test/packet_sender.h"
-
 #include <algorithm>
 
+#include "webrtc/modules/remote_bitrate_estimator/test/packet_sender.h"
+
 namespace webrtc {
 namespace testing {
 namespace bwe {

webrtc/modules/remote_bitrate_estimator/test/packet_sender.h

@@ -13,6 +13,7 @@
 
 #include <list>
 #include <limits>
+#include <set>
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
@@ -149,7 +150,7 @@ class TcpSender : public PacketSender {
  private:
   struct InFlight {
    public:
-    InFlight(const MediaPacket& packet)
+    explicit InFlight(const MediaPacket& packet)
         : sequence_number(packet.header().sequenceNumber),
           time_ms(packet.send_time_ms()) {}
 

webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc

@@ -10,8 +10,10 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
 
-#include <sstream>
 #include <stdio.h>
+
+#include <set>
+#include <sstream>
 #include <string>
 
 #include "gflags/gflags.h"