Use newer version of TimeDelta and TimeStamp factories in modules/

This change generated with following commands:
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format

Bug: None
Change-Id: I117d64a54950be040d996035c54bc0043310943a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168340
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30489}

4 files changed, 23 insertions, 23 deletions

diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
index 606237d303..70a50d6de7 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
@@ -178,7 +178,7 @@ TEST(AudioNetworkAdaptorImplTest,
       "WebRTC-Audio-BitrateAdaptation/Enabled/WebRTC-Audio-FecAdaptation/"
       "Enabled/");
   rtc::ScopedFakeClock fake_clock;
-  fake_clock.AdvanceTime(TimeDelta::ms(kClockInitialTimeMs));
+  fake_clock.AdvanceTime(TimeDelta::Millis(kClockInitialTimeMs));
   auto states = CreateAudioNetworkAdaptor();
   AudioEncoderRuntimeConfig config;
   config.bitrate_bps = 32000;
@@ -196,7 +196,7 @@ TEST(AudioNetworkAdaptorImplTest,
 TEST(AudioNetworkAdaptorImplTest,
      DumpNetworkMetricsIsCalledOnSetNetworkMetrics) {
   rtc::ScopedFakeClock fake_clock;
-  fake_clock.AdvanceTime(TimeDelta::ms(kClockInitialTimeMs));
+  fake_clock.AdvanceTime(TimeDelta::Millis(kClockInitialTimeMs));
 
   auto states = CreateAudioNetworkAdaptor();
 
@@ -214,31 +214,31 @@ TEST(AudioNetworkAdaptorImplTest,
               DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
   states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
 
-  fake_clock.AdvanceTime(TimeDelta::ms(100));
+  fake_clock.AdvanceTime(TimeDelta::Millis(100));
   timestamp_check += 100;
   check.uplink_packet_loss_fraction = kPacketLoss;
   EXPECT_CALL(*states.mock_debug_dump_writer,
               DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
   states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
 
-  fake_clock.AdvanceTime(TimeDelta::ms(50));
+  fake_clock.AdvanceTime(TimeDelta::Millis(50));
   timestamp_check += 50;
 
-  fake_clock.AdvanceTime(TimeDelta::ms(200));
+  fake_clock.AdvanceTime(TimeDelta::Millis(200));
   timestamp_check += 200;
   check.rtt_ms = kRtt;
   EXPECT_CALL(*states.mock_debug_dump_writer,
               DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
   states.audio_network_adaptor->SetRtt(kRtt);
 
-  fake_clock.AdvanceTime(TimeDelta::ms(150));
+  fake_clock.AdvanceTime(TimeDelta::Millis(150));
   timestamp_check += 150;
   check.target_audio_bitrate_bps = kTargetAudioBitrate;
   EXPECT_CALL(*states.mock_debug_dump_writer,
               DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
   states.audio_network_adaptor->SetTargetAudioBitrate(kTargetAudioBitrate);
 
-  fake_clock.AdvanceTime(TimeDelta::ms(50));
+  fake_clock.AdvanceTime(TimeDelta::Millis(50));
   timestamp_check += 50;
   check.overhead_bytes_per_packet = kOverhead;
   EXPECT_CALL(*states.mock_debug_dump_writer,
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
index c267b791c3..4286434b5b 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
@@ -147,7 +147,7 @@ TEST(ControllerManagerTest, DoNotReorderBeforeMinReordingTime) {
   CheckControllersOrder(&states, kChracteristicBandwithBps[0],
                         kChracteristicPacketLossFraction[0],
                         {kNumControllers - 2, kNumControllers - 1, 0, 1});
-  fake_clock.AdvanceTime(TimeDelta::ms(kMinReorderingTimeMs - 1));
+  fake_clock.AdvanceTime(TimeDelta::Millis(kMinReorderingTimeMs - 1));
   // Move uplink bandwidth and packet loss fraction to the other controller's
   // characteristic point, which would cause controller manager to reorder the
   // controllers if time had reached min reordering time.
@@ -168,7 +168,7 @@ TEST(ControllerManagerTest, ReorderBeyondMinReordingTimeAndMinDistance) {
   // of two controllers.
   CheckControllersOrder(&states, kBandwidthBps, kPacketLossFraction,
                         {kNumControllers - 2, kNumControllers - 1, 0, 1});
-  fake_clock.AdvanceTime(TimeDelta::ms(kMinReorderingTimeMs));
+  fake_clock.AdvanceTime(TimeDelta::Millis(kMinReorderingTimeMs));
   // Then let network metrics move a little towards the other controller.
   CheckControllersOrder(&states, kBandwidthBps - kMinBandwithChangeBps - 1,
                         kPacketLossFraction,
@@ -187,7 +187,7 @@ TEST(ControllerManagerTest, DoNotReorderIfNetworkMetricsChangeTooSmall) {
   // of two controllers.
   CheckControllersOrder(&states, kBandwidthBps, kPacketLossFraction,
                         {kNumControllers - 2, kNumControllers - 1, 0, 1});
-  fake_clock.AdvanceTime(TimeDelta::ms(kMinReorderingTimeMs));
+  fake_clock.AdvanceTime(TimeDelta::Millis(kMinReorderingTimeMs));
   // Then let network metrics move a little towards the other controller.
   CheckControllersOrder(&states, kBandwidthBps - kMinBandwithChangeBps + 1,
                         kPacketLossFraction,
@@ -349,7 +349,7 @@ TEST(ControllerManagerTest, DebugDumpLoggedWhenCreateFromConfigString) {
 
   constexpr int64_t kClockInitialTimeMs = 12345678;
   rtc::ScopedFakeClock fake_clock;
-  fake_clock.AdvanceTime(TimeDelta::ms(kClockInitialTimeMs));
+  fake_clock.AdvanceTime(TimeDelta::Millis(kClockInitialTimeMs));
   auto debug_dump_writer =
       std::unique_ptr<MockDebugDumpWriter>(new NiceMock<MockDebugDumpWriter>());
   EXPECT_CALL(*debug_dump_writer, Die());
@@ -446,7 +446,7 @@ TEST(ControllerManagerTest, CreateFromConfigStringAndCheckReordering) {
 
   metrics.uplink_bandwidth_bps = kChracteristicBandwithBps[1];
   metrics.uplink_packet_loss_fraction = kChracteristicPacketLossFraction[1];
-  fake_clock.AdvanceTime(TimeDelta::ms(kMinReorderingTimeMs - 1));
+  fake_clock.AdvanceTime(TimeDelta::Millis(kMinReorderingTimeMs - 1));
   controllers = states.controller_manager->GetSortedControllers(metrics);
   // Should not reorder since min reordering time is not met.
   CheckControllersOrder(controllers,
@@ -455,7 +455,7 @@ TEST(ControllerManagerTest, CreateFromConfigStringAndCheckReordering) {
                             ControllerType::CHANNEL, ControllerType::DTX,
                             ControllerType::BIT_RATE});
 
-  fake_clock.AdvanceTime(TimeDelta::ms(1));
+  fake_clock.AdvanceTime(TimeDelta::Millis(1));
   controllers = states.controller_manager->GetSortedControllers(metrics);
   // Reorder now.
   CheckControllersOrder(controllers,
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 168bcec241..768031b0fb 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -924,11 +924,11 @@ AudioEncoderOpusImpl::GetFrameLengthRange() const {
   if (config_.supported_frame_lengths_ms.empty()) {
     return absl::nullopt;
   } else if (audio_network_adaptor_) {
-    return {{TimeDelta::ms(config_.supported_frame_lengths_ms.front()),
-             TimeDelta::ms(config_.supported_frame_lengths_ms.back())}};
+    return {{TimeDelta::Millis(config_.supported_frame_lengths_ms.front()),
+             TimeDelta::Millis(config_.supported_frame_lengths_ms.back())}};
   } else {
-    return {{TimeDelta::ms(config_.frame_size_ms),
-             TimeDelta::ms(config_.frame_size_ms)}};
+    return {{TimeDelta::Millis(config_.frame_size_ms),
+             TimeDelta::Millis(config_.frame_size_ms)}};
   }
 }
 
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index 2192f40643..fd8e7e2665 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -57,7 +57,7 @@ std::unique_ptr<AudioEncoderOpusStates> CreateCodec(int sample_rate_hz,
       std::make_unique<AudioEncoderOpusStates>();
   states->mock_audio_network_adaptor = nullptr;
   states->fake_clock.reset(new rtc::ScopedFakeClock());
-  states->fake_clock->SetTime(Timestamp::us(kInitialTimeUs));
+  states->fake_clock->SetTime(Timestamp::Micros(kInitialTimeUs));
 
   MockAudioNetworkAdaptor** mock_ptr = &states->mock_audio_network_adaptor;
   AudioEncoderOpusImpl::AudioNetworkAdaptorCreator creator =
@@ -249,7 +249,7 @@ void TestSetPacketLossRate(const AudioEncoderOpusStates* states,
   constexpr int64_t kSampleIntervalMs = 184198;
   for (float loss : losses) {
     states->encoder->OnReceivedUplinkPacketLossFraction(loss);
-    states->fake_clock->AdvanceTime(TimeDelta::ms(kSampleIntervalMs));
+    states->fake_clock->AdvanceTime(TimeDelta::Millis(kSampleIntervalMs));
     EXPECT_FLOAT_EQ(expected_return, states->encoder->packet_loss_rate());
   }
 }
@@ -429,7 +429,7 @@ TEST_P(AudioEncoderOpusTest,
   states->encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_1);
   EXPECT_FLOAT_EQ(0.01f, states->encoder->packet_loss_rate());
 
-  states->fake_clock->AdvanceTime(TimeDelta::ms(kSecondSampleTimeMs));
+  states->fake_clock->AdvanceTime(TimeDelta::Millis(kSecondSampleTimeMs));
   states->encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_2);
 
   // Now the output of packet loss fraction smoother should be
@@ -667,8 +667,8 @@ TEST_P(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
   // Repeat update uplink bandwidth tests.
   for (int i = 0; i < 5; i++) {
     // Don't update till it is time to update again.
-    states->fake_clock->AdvanceTime(
-        TimeDelta::ms(states->config.uplink_bandwidth_update_interval_ms - 1));
+    states->fake_clock->AdvanceTime(TimeDelta::Millis(
+        states->config.uplink_bandwidth_update_interval_ms - 1));
     states->encoder->Encode(
         0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
 
@@ -676,7 +676,7 @@ TEST_P(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
     EXPECT_CALL(*states->mock_bitrate_smoother, GetAverage())
         .WillOnce(Return(40000));
     EXPECT_CALL(*states->mock_audio_network_adaptor, SetUplinkBandwidth(40000));
-    states->fake_clock->AdvanceTime(TimeDelta::ms(1));
+    states->fake_clock->AdvanceTime(TimeDelta::Millis(1));
     states->encoder->Encode(
         0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
   }