commit 5f83cf0c6d842352d8e017321f0223aefebb1bef
author Sebastian Jansson <srte@webrtc.org> Tue May 08 14:52:22 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Tue May 08 13:22:53 2018 +0000
tree 3a82358adf0e282e0c0de4c58ab85c22a0d1409e
parent 5b2b692079e65be33998e6b8e7e7b7efce3cf9bf

Replacing rtc::TimeDelta with webrtc::TimeDelta.

This removes the redundant type and replaces all usages. A slight change
in behavior is that we no longer get nanosecond resolution. This should
not matter since no current code requires nanosecond resolution.

Bug: webrtc:9155
Change-Id: I04334e08c686d95731621a6c8a7e40400d0ae3b2
Reviewed-on: https://webrtc-review.googlesource.com/71163
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23174}

modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc

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 574b699..2872219 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
@@ -192,7 +192,7 @@
       "WebRTC-Audio-BitrateAdaptation/Enabled/WebRTC-Audio-FecAdaptation/"
       "Enabled/");
   rtc::ScopedFakeClock fake_clock;
-  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(kClockInitialTimeMs));
+  fake_clock.AdvanceTime(TimeDelta::ms(kClockInitialTimeMs));
   auto states = CreateAudioNetworkAdaptor();
   AudioEncoderRuntimeConfig config;
   config.bitrate_bps = 32000;
@@ -210,7 +210,7 @@
 TEST(AudioNetworkAdaptorImplTest,
      DumpNetworkMetricsIsCalledOnSetNetworkMetrics) {
   rtc::ScopedFakeClock fake_clock;
-  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(kClockInitialTimeMs));
+  fake_clock.AdvanceTime(TimeDelta::ms(kClockInitialTimeMs));
 
   auto states = CreateAudioNetworkAdaptor();
 
@@ -229,14 +229,14 @@
               DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
   states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
 
-  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(100));
+  fake_clock.AdvanceTime(TimeDelta::ms(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(rtc::TimeDelta::FromMilliseconds(50));
+  fake_clock.AdvanceTime(TimeDelta::ms(50));
   timestamp_check += 50;
   check.uplink_recoverable_packet_loss_fraction = kRecoverablePacketLoss;
   EXPECT_CALL(*states.mock_debug_dump_writer,
@@ -244,21 +244,21 @@
   states.audio_network_adaptor->SetUplinkRecoverablePacketLossFraction(
       kRecoverablePacketLoss);
 
-  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(200));
+  fake_clock.AdvanceTime(TimeDelta::ms(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(rtc::TimeDelta::FromMilliseconds(150));
+  fake_clock.AdvanceTime(TimeDelta::ms(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(rtc::TimeDelta::FromMilliseconds(50));
+  fake_clock.AdvanceTime(TimeDelta::ms(50));
   timestamp_check += 50;
   check.overhead_bytes_per_packet = kOverhead;
   EXPECT_CALL(*states.mock_debug_dump_writer,