Removing warning suppression flags from pc/.

Bug: webrtc:9251
Change-Id: Ic12126fc03309448fe71a17e6b65343949496f4f
Reviewed-on: https://webrtc-review.googlesource.com/86820
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23838}

pc/BUILD.gn

@@ -112,15 +112,6 @@ rtc_source_set("rtc_pc") {
   ]
 }
 
-config("libjingle_peerconnection_warnings_config") {
-  # GN orders flags on a target before flags from configs. The default config
-  # adds these flags so to cancel them out they need to come from a config and
-  # cannot be on the target directly.
-  if (!is_win && !is_clang) {
-    cflags = [ "-Wno-maybe-uninitialized" ]  # Only exists for GCC.
-  }
-}
-
 rtc_static_library("peerconnection") {
   visibility = [ "*" ]
   cflags = []
@@ -182,8 +173,6 @@ rtc_static_library("peerconnection") {
     "webrtcsessiondescriptionfactory.h",
   ]
 
-  configs += [ ":libjingle_peerconnection_warnings_config" ]
-
   if (!build_with_chromium && is_clang) {
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
@@ -249,8 +238,6 @@ rtc_static_library("create_pc_factory") {
     "../rtc_base:rtc_base_approved",
   ]
 
-  configs += [ ":libjingle_peerconnection_warnings_config" ]
-
   if (!build_with_chromium && is_clang) {
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
@@ -271,15 +258,6 @@ rtc_source_set("libjingle_peerconnection") {
 }
 
 if (rtc_include_tests) {
-  config("rtc_pc_unittests_config") {
-    # GN orders flags on a target before flags from configs. The default config
-    # adds -Wall, and this flag have to be after -Wall -- so they need to
-    # come from a config and can't be on the target directly.
-    if (!is_win && !is_clang) {
-      cflags = [ "-Wno-maybe-uninitialized" ]  # Only exists for GCC.
-    }
-  }
-
   rtc_test("rtc_pc_unittests") {
     testonly = true
 
@@ -301,8 +279,6 @@ if (rtc_include_tests) {
 
     include_dirs = [ "//third_party/libsrtp/srtp" ]
 
-    configs += [ ":rtc_pc_unittests_config" ]
-
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
@@ -440,25 +416,6 @@ if (rtc_include_tests) {
     }
   }
 
-  config("peerconnection_unittests_config") {
-    # The warnings below are enabled by default. Since GN orders compiler flags
-    # for a target before flags from configs, the only way to disable such
-    # warnings is by having them in a separate config, loaded from the target.
-    # TODO(kjellander): Make the code compile without disabling these flags.
-    # See https://bugs.webrtc.org/3307.
-    if (is_clang && is_win) {
-      cflags = [
-        # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6267
-        # for -Wno-sign-compare
-        "-Wno-sign-compare",
-      ]
-    }
-
-    if (!is_win) {
-      cflags = [ "-Wno-sign-compare" ]
-    }
-  }
-
   rtc_test("peerconnection_unittests") {
     testonly = true
     sources = [
@@ -505,8 +462,6 @@ if (rtc_include_tests) {
       defines = [ "HAVE_SCTP" ]
     }
 
-    configs += [ ":peerconnection_unittests_config" ]
-
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
@@ -562,6 +517,7 @@ if (rtc_include_tests) {
       "../rtc_base:rtc_base_tests_main",
       "../rtc_base:rtc_base_tests_utils",
       "../rtc_base:rtc_task_queue",
+      "../rtc_base:safe_conversions",
       "../system_wrappers:metrics_default",
       "../system_wrappers:runtime_enabled_features_default",
       "../test:audio_codec_mocks",

pc/datachannel_unittest.cc

@@ -15,6 +15,7 @@
 #include "pc/sctputils.h"
 #include "pc/test/fakedatachannelprovider.h"
 #include "rtc_base/gunit.h"
+#include "rtc_base/numerics/safe_conversions.h"
 
 using webrtc::DataChannel;
 using webrtc::SctpSidAllocator;
@@ -172,7 +173,8 @@ TEST_F(SctpDataChannelTest, BufferedAmountWhenBlocked) {
   }
   EXPECT_EQ(buffer.data.size() * number_of_packets,
             webrtc_data_channel_->buffered_amount());
-  EXPECT_EQ(number_of_packets, observer_->on_buffered_amount_change_count());
+  EXPECT_EQ(rtc::checked_cast<size_t>(number_of_packets),
+            observer_->on_buffered_amount_change_count());
 }
 
 // Tests that the queued data are sent when the channel transitions from blocked

pc/peerconnection_integrationtest.cc

@@ -60,6 +60,7 @@
 #include "rtc_base/fakenetwork.h"
 #include "rtc_base/firewallsocketserver.h"
 #include "rtc_base/gunit.h"
+#include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/testcertificateverifier.h"
 #include "rtc_base/virtualsocketserver.h"
 #include "test/gmock.h"
@@ -1595,8 +1596,8 @@ TEST_P(PeerConnectionIntegrationTest,
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Should be one receiver each for audio/video.
-  EXPECT_EQ(2, caller()->rtp_receiver_observers().size());
+  EXPECT_EQ(2U, caller()->rtp_receiver_observers().size());
-  EXPECT_EQ(2, callee()->rtp_receiver_observers().size());
+  EXPECT_EQ(2U, callee()->rtp_receiver_observers().size());
   // Wait for all "first packet received" callbacks to be fired.
   EXPECT_TRUE_WAIT(
       std::all_of(caller()->rtp_receiver_observers().begin(),
@@ -1616,8 +1617,8 @@ TEST_P(PeerConnectionIntegrationTest,
   // callback should still be invoked.
   caller()->ResetRtpReceiverObservers();
   callee()->ResetRtpReceiverObservers();
-  EXPECT_EQ(2, caller()->rtp_receiver_observers().size());
+  EXPECT_EQ(2U, caller()->rtp_receiver_observers().size());
-  EXPECT_EQ(2, callee()->rtp_receiver_observers().size());
+  EXPECT_EQ(2U, callee()->rtp_receiver_observers().size());
   EXPECT_TRUE(
       std::all_of(caller()->rtp_receiver_observers().begin(),
                   caller()->rtp_receiver_observers().end(),
@@ -1903,7 +1904,7 @@ TEST_P(PeerConnectionIntegrationTest, AudioToVideoUpgrade) {
       // The caller creates a new transceiver to receive video on when receiving
       // the offer, but by default it is send only.
       auto transceivers = caller()->pc()->GetTransceivers();
-      ASSERT_EQ(3, transceivers.size());
+      ASSERT_EQ(3U, transceivers.size());
       ASSERT_EQ(cricket::MEDIA_TYPE_VIDEO,
                 transceivers[2]->receiver()->media_type());
       transceivers[2]->sender()->SetTrack(caller()->CreateLocalVideoTrack());
@@ -2590,7 +2591,7 @@ TEST_P(PeerConnectionIntegrationTest,
   //
   // Also, we use "EXPECT_TRUE_WAIT" because the stats collector may decide to
   // return cached stats if not enough time has passed since the last update.
-  EXPECT_TRUE_WAIT(callee()->OldGetStats()->BytesReceived() > 0U,
+  EXPECT_TRUE_WAIT(callee()->OldGetStats()->BytesReceived() > 0,
                    kDefaultTimeout);
 }
 
@@ -3209,10 +3210,10 @@ TEST_P(PeerConnectionIntegrationTest, StressTestUnorderedSctpDataChannel) {
   }
 
   // Wait for all messages to be received.
-  EXPECT_EQ_WAIT(kNumMessages,
+  EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
                  caller()->data_observer()->received_message_count(),
                  kDefaultTimeout);
-  EXPECT_EQ_WAIT(kNumMessages,
+  EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
                  callee()->data_observer()->received_message_count(),
                  kDefaultTimeout);
 
@@ -3517,17 +3518,17 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyBestConnection) {
   if (TestIPv6()) {
     // When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4
     // connection.
-    EXPECT_EQ(0u, num_best_ipv4);
+    EXPECT_EQ(0, num_best_ipv4);
-    EXPECT_EQ(1u, num_best_ipv6);
+    EXPECT_EQ(1, num_best_ipv6);
   } else {
-    EXPECT_EQ(1u, num_best_ipv4);
+    EXPECT_EQ(1, num_best_ipv4);
-    EXPECT_EQ(0u, num_best_ipv6);
+    EXPECT_EQ(0, num_best_ipv6);
   }
 
-  EXPECT_EQ(0u, metrics_observer->GetEnumCounter(
+  EXPECT_EQ(0, metrics_observer->GetEnumCounter(
                     webrtc::kEnumCounterIceCandidatePairTypeUdp,
                     webrtc::kIceCandidatePairHostHost));
-  EXPECT_EQ(1u, metrics_observer->GetEnumCounter(
+  EXPECT_EQ(1, metrics_observer->GetEnumCounter(
                     webrtc::kEnumCounterIceCandidatePairTypeUdp,
                     webrtc::kIceCandidatePairHostPublicHostPublic));
 }
@@ -3829,7 +3830,7 @@ TEST_F(PeerConnectionIntegrationTestPlanB, CanSendRemoteVideoTrack) {
   caller()->AddVideoTrack();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
-  ASSERT_EQ(1, callee()->remote_streams()->count());
+  ASSERT_EQ(1U, callee()->remote_streams()->count());
 
   // Echo the stream back, and do a new offer/anwer (initiated by callee this
   // time).

pc/peerconnectionendtoend_unittest.cc

@@ -459,8 +459,8 @@ TEST_P(PeerConnectionEndToEndTest, CallWithCustomCodec) {
   // the same ID because they were passed to the same PeerConnectionFactory,
   // and the second pair got the same ID---but these two IDs are not equal,
   // because each PeerConnectionFactory has its own ID.
-  EXPECT_EQ(1, encoder_id1.size());
+  EXPECT_EQ(1U, encoder_id1.size());
-  EXPECT_EQ(1, encoder_id2.size());
+  EXPECT_EQ(1U, encoder_id2.size());
   EXPECT_EQ(encoder_id1, decoder_id1);
   EXPECT_EQ(encoder_id2, decoder_id2);
   EXPECT_NE(encoder_id1, encoder_id2);
@@ -540,16 +540,16 @@ TEST_P(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
   Negotiate();
   WaitForConnection();
 
-  EXPECT_EQ(1U, caller_dc_1->id() % 2);
+  EXPECT_EQ(1, caller_dc_1->id() % 2);
-  EXPECT_EQ(0U, callee_dc_1->id() % 2);
+  EXPECT_EQ(0, callee_dc_1->id() % 2);
 
   rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
       caller_->CreateDataChannel("data", init));
   rtc::scoped_refptr<DataChannelInterface> callee_dc_2(
       callee_->CreateDataChannel("data", init));
 
-  EXPECT_EQ(1U, caller_dc_2->id() % 2);
+  EXPECT_EQ(1, caller_dc_2->id() % 2);
-  EXPECT_EQ(0U, callee_dc_2->id() % 2);
+  EXPECT_EQ(0, callee_dc_2->id() % 2);
 }
 
 // Verifies that the message is received by the right remote DataChannel when

pc/peerconnectioninterface_unittest.cc

@@ -3408,13 +3408,13 @@ TEST_P(PeerConnectionInterfaceTest, SetConfigurationCausingPartialIceRestart) {
 
   // Grab the ufrags.
   std::vector<std::string> initial_ufrags = GetUfrags(pc_->local_description());
-  ASSERT_EQ(2, initial_ufrags.size());
+  ASSERT_EQ(2U, initial_ufrags.size());
 
   // Create offer and grab the new ufrags.
   CreateOfferAsLocalDescription();
   std::vector<std::string> subsequent_ufrags =
       GetUfrags(pc_->local_description());
-  ASSERT_EQ(2, subsequent_ufrags.size());
+  ASSERT_EQ(2U, subsequent_ufrags.size());
 
   // Ensure that only the ufrag for the second m= section changed.
   EXPECT_EQ(initial_ufrags[0], subsequent_ufrags[0]);

pc/rtcstats_integrationtest.cc

@@ -343,7 +343,7 @@ class RTCStatsReportVerifier {
     bool verify_successful = true;
     std::vector<const RTCTransportStats*> transport_stats =
         report_->GetStatsOfType<RTCTransportStats>();
-    EXPECT_EQ(transport_stats.size(), 1);
+    EXPECT_EQ(transport_stats.size(), 1U);
     std::string selected_candidate_pair_id =
         *transport_stats[0]->selected_candidate_pair_id;
     for (const RTCStats& stats : *report_) {

pc/rtcstatscollector_unittest.cc

@@ -110,7 +110,7 @@ template <typename T>
 std::string IdForType(const RTCStatsReport* report) {
   auto stats_of_my_type = report->RTCStatsReport::GetStatsOfType<T>();
   // We cannot use ASSERT here, since we're within a function.
-  EXPECT_EQ(1, stats_of_my_type.size())
+  EXPECT_EQ(1U, stats_of_my_type.size())
       << "Unexpected number of stats of this type";
   if (stats_of_my_type.size() == 1) {
     return stats_of_my_type[0]->id();
@@ -1470,9 +1470,9 @@ TEST_F(RTCStatsCollectorTest,
   rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
 
   auto stats_of_my_type = report->GetStatsOfType<RTCMediaStreamStats>();
-  ASSERT_EQ(1, stats_of_my_type.size()) << "No stream in " << report->ToJson();
+  ASSERT_EQ(1U, stats_of_my_type.size()) << "No stream in " << report->ToJson();
   auto stats_of_track_type = report->GetStatsOfType<RTCMediaStreamTrackStats>();
-  ASSERT_EQ(1, stats_of_track_type.size())
+  ASSERT_EQ(1U, stats_of_track_type.size())
       << "Wrong number of tracks in " << report->ToJson();
 
   RTCMediaStreamStats expected_local_stream(stats_of_my_type[0]->id(),
@@ -1533,9 +1533,9 @@ TEST_F(RTCStatsCollectorTest,
   rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
 
   auto stats_of_my_type = report->GetStatsOfType<RTCMediaStreamStats>();
-  ASSERT_EQ(1, stats_of_my_type.size()) << "No stream in " << report->ToJson();
+  ASSERT_EQ(1U, stats_of_my_type.size()) << "No stream in " << report->ToJson();
   auto stats_of_track_type = report->GetStatsOfType<RTCMediaStreamTrackStats>();
-  ASSERT_EQ(1, stats_of_track_type.size())
+  ASSERT_EQ(1U, stats_of_track_type.size())
       << "Wrong number of tracks in " << report->ToJson();
   ASSERT_TRUE(*(stats_of_track_type[0]->remote_source));
 
@@ -1598,7 +1598,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
   rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
 
   auto stats_of_track_type = report->GetStatsOfType<RTCMediaStreamTrackStats>();
-  ASSERT_EQ(1, stats_of_track_type.size());
+  ASSERT_EQ(1U, stats_of_track_type.size());
 
   RTCInboundRTPStreamStats expected_audio("RTCInboundRTPAudioStream_1",
                                           report->timestamp_us());
@@ -1775,9 +1775,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
   rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
 
   auto stats_of_my_type = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
-  ASSERT_EQ(1, stats_of_my_type.size());
+  ASSERT_EQ(1U, stats_of_my_type.size());
   auto stats_of_track_type = report->GetStatsOfType<RTCMediaStreamTrackStats>();
-  ASSERT_EQ(1, stats_of_track_type.size());
+  ASSERT_EQ(1U, stats_of_track_type.size());
 
   RTCOutboundRTPStreamStats expected_video(stats_of_my_type[0]->id(),
                                            report->timestamp_us());
@@ -2086,11 +2086,11 @@ TEST_F(RTCStatsCollectorTest, StatsReportedOnZeroSsrc) {
 
   std::vector<const RTCMediaStreamTrackStats*> track_stats =
       report->GetStatsOfType<RTCMediaStreamTrackStats>();
-  EXPECT_EQ(1, track_stats.size());
+  EXPECT_EQ(1U, track_stats.size());
 
   std::vector<const RTCRTPStreamStats*> rtp_stream_stats =
       report->GetStatsOfType<RTCRTPStreamStats>();
-  EXPECT_EQ(0, rtp_stream_stats.size());
+  EXPECT_EQ(0U, rtp_stream_stats.size());
 }
 
 TEST_F(RTCStatsCollectorTest, DoNotCrashOnSsrcChange) {
@@ -2106,7 +2106,7 @@ TEST_F(RTCStatsCollectorTest, DoNotCrashOnSsrcChange) {
 
   std::vector<const RTCMediaStreamTrackStats*> track_stats =
       report->GetStatsOfType<RTCMediaStreamTrackStats>();
-  EXPECT_EQ(1, track_stats.size());
+  EXPECT_EQ(1U, track_stats.size());
 }
 
 // Used for test below, to test calling GetStatsReport during a callback.

pc/rtpsenderreceiver_unittest.cc

@@ -651,7 +651,7 @@ TEST_F(RtpSenderReceiverTest, AudioSenderCheckTransactionIdRefresh) {
   CreateAudioRtpSender();
 
   RtpParameters params = audio_rtp_sender_->GetParameters();
-  EXPECT_NE(params.transaction_id.size(), 0);
+  EXPECT_NE(params.transaction_id.size(), 0U);
   auto saved_transaction_id = params.transaction_id;
   params = audio_rtp_sender_->GetParameters();
   EXPECT_NE(saved_transaction_id, params.transaction_id);
@@ -751,19 +751,19 @@ TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) {
 
   EXPECT_EQ(-1, voice_media_channel_->max_bps());
   webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
-  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1U, params.encodings.size());
   EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
   params.encodings[0].max_bitrate_bps = 1000;
   EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
 
   // Read back the parameters and verify they have been changed.
   params = audio_rtp_sender_->GetParameters();
-  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1U, params.encodings.size());
   EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
 
   // Verify that the audio channel received the new parameters.
   params = voice_media_channel_->GetRtpSendParameters(kAudioSsrc);
-  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1U, params.encodings.size());
   EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
 
   // Verify that the global bitrate limit has not been changed.
@@ -776,7 +776,7 @@ TEST_F(RtpSenderReceiverTest, SetAudioBitratePriority) {
   CreateAudioRtpSender();
 
   webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
-  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1U, params.encodings.size());
   EXPECT_EQ(webrtc::kDefaultBitratePriority,
             params.encodings[0].bitrate_priority);
   double new_bitrate_priority = 2.0;
@@ -784,11 +784,11 @@ TEST_F(RtpSenderReceiverTest, SetAudioBitratePriority) {
   EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
 
   params = audio_rtp_sender_->GetParameters();
-  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1U, params.encodings.size());
   EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
 
   params = voice_media_channel_->GetRtpSendParameters(kAudioSsrc);
-  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1U, params.encodings.size());
   EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
 
   DestroyAudioRtpSender();
@@ -843,7 +843,7 @@ TEST_F(RtpSenderReceiverTest, VideoSenderCheckTransactionIdRefresh) {
   CreateVideoRtpSender();
 
   RtpParameters params = video_rtp_sender_->GetParameters();
-  EXPECT_NE(params.transaction_id.size(), 0);
+  EXPECT_NE(params.transaction_id.size(), 0U);
   auto saved_transaction_id = params.transaction_id;
   params = video_rtp_sender_->GetParameters();
   EXPECT_NE(saved_transaction_id, params.transaction_id);
@@ -968,7 +968,7 @@ TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrate) {
 
   EXPECT_EQ(-1, video_media_channel_->max_bps());
   webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
-  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1U, params.encodings.size());
   EXPECT_FALSE(params.encodings[0].min_bitrate_bps);
   EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
   params.encodings[0].min_bitrate_bps = 100;
@@ -977,13 +977,13 @@ TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrate) {
 
   // Read back the parameters and verify they have been changed.
   params = video_rtp_sender_->GetParameters();
-  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1U, params.encodings.size());
   EXPECT_EQ(100, params.encodings[0].min_bitrate_bps);
   EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
 
   // Verify that the video channel received the new parameters.
   params = video_media_channel_->GetRtpSendParameters(kVideoSsrc);
-  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1U, params.encodings.size());
   EXPECT_EQ(100, params.encodings[0].min_bitrate_bps);
   EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
 
@@ -1025,7 +1025,7 @@ TEST_F(RtpSenderReceiverTest, SetVideoBitratePriority) {
   CreateVideoRtpSender();
 
   webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
-  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1U, params.encodings.size());
   EXPECT_EQ(webrtc::kDefaultBitratePriority,
             params.encodings[0].bitrate_priority);
   double new_bitrate_priority = 2.0;
@@ -1033,11 +1033,11 @@ TEST_F(RtpSenderReceiverTest, SetVideoBitratePriority) {
   EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
 
   params = video_rtp_sender_->GetParameters();
-  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1U, params.encodings.size());
   EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
 
   params = video_media_channel_->GetRtpSendParameters(kVideoSsrc);
-  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1U, params.encodings.size());
   EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
 
   DestroyVideoRtpSender();

pc/webrtcsdp_unittest.cc

@@ -3544,7 +3544,7 @@ TEST_F(WebRtcSdpTest, IceCredentialsInCandidateStringIgnored) {
       SdpDeserialize(kSdpWithIceCredentialsInCandidateString, &jdesc_output));
   const IceCandidateCollection* candidates = jdesc_output.candidates(0);
   ASSERT_NE(nullptr, candidates);
-  ASSERT_EQ(1, candidates->count());
+  ASSERT_EQ(1U, candidates->count());
   cricket::Candidate c = candidates->at(0)->candidate();
   EXPECT_EQ("ufrag_voice", c.username());
   EXPECT_EQ("pwd_voice", c.password());