commit 6a489f22c71bc332886f76b493eead7551fc17fe
author Mirko Bonadei <mbonadei@webrtc.org> Tue Apr 09 13:11:12 2019
committer Commit Bot <commit-bot@chromium.org> Tue Apr 09 17:18:20 2019
tree d6c7fd756e1b0d992070f1944bd5e9a372edf242
parent a857698d549df4977d38555af220e7839e484cfc

Fully qualify googletest symbols.

Semi-automatically created with:

git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format

After this, two .cc files failed to compile and I have fixed them
manually.

Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}

audio/audio_receive_stream_unittest.cc

diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index def4d1a..127f2ff 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -33,9 +33,9 @@
 namespace test {
 namespace {
 
-using testing::_;
-using testing::FloatEq;
-using testing::Return;
+using ::testing::_;
+using ::testing::FloatEq;
+using ::testing::Return;
 
 AudioDecodingCallStats MakeAudioDecodeStatsForTest() {
   AudioDecodingCallStats audio_decode_stats;
@@ -76,7 +76,7 @@
 
   explicit ConfigHelper(rtc::scoped_refptr<MockAudioMixer> audio_mixer)
       : audio_mixer_(audio_mixer) {
-    using testing::Invoke;
+    using ::testing::Invoke;
 
     AudioState::Config config;
     config.audio_mixer = audio_mixer_;
@@ -85,7 +85,7 @@
         new rtc::RefCountedObject<testing::NiceMock<MockAudioDeviceModule>>();
     audio_state_ = AudioState::Create(config);
 
-    channel_receive_ = new testing::StrictMock<MockChannelReceive>();
+    channel_receive_ = new ::testing::StrictMock<MockChannelReceive>();
     EXPECT_CALL(*channel_receive_, SetLocalSSRC(kLocalSsrc)).Times(1);
     EXPECT_CALL(*channel_receive_, SetNACKStatus(true, 15)).Times(1);
     EXPECT_CALL(*channel_receive_,
@@ -96,7 +96,7 @@
     EXPECT_CALL(*channel_receive_, SetAssociatedSendChannel(nullptr)).Times(1);
     EXPECT_CALL(*channel_receive_, SetReceiveCodecs(_))
         .WillRepeatedly(Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
-          EXPECT_THAT(codecs, testing::IsEmpty());
+          EXPECT_THAT(codecs, ::testing::IsEmpty());
         }));
 
     stream_config_.rtp.local_ssrc = kLocalSsrc;
@@ -124,8 +124,8 @@
   MockChannelReceive* channel_receive() { return channel_receive_; }
 
   void SetupMockForGetStats() {
-    using testing::DoAll;
-    using testing::SetArgPointee;
+    using ::testing::DoAll;
+    using ::testing::SetArgPointee;
 
     ASSERT_TRUE(channel_receive_);
     EXPECT_CALL(*channel_receive_, GetRTCPStatistics())
@@ -152,7 +152,7 @@
   rtc::scoped_refptr<AudioState> audio_state_;
   rtc::scoped_refptr<MockAudioMixer> audio_mixer_;
   AudioReceiveStream::Config stream_config_;
-  testing::StrictMock<MockChannelReceive>* channel_receive_ = nullptr;
+  ::testing::StrictMock<MockChannelReceive>* channel_receive_ = nullptr;
   RtpStreamReceiverController rtp_stream_receiver_controller_;
   MockTransport rtcp_send_transport_;
 };
@@ -242,7 +242,7 @@
   parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);
 
   EXPECT_CALL(*helper.channel_receive(),
-              OnRtpPacket(testing::Ref(parsed_packet)));
+              OnRtpPacket(::testing::Ref(parsed_packet)));
 
   recv_stream->OnRtpPacket(parsed_packet);
 }