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}

3 files changed, 6 insertions, 5 deletions

diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 3f00dfaed4..7eee652411 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -485,6 +485,7 @@ if (rtc_include_tests) {
       ":rtc_pc_base",
       "../api:libjingle_peerconnection_api",
       "../api:mock_rtp",
+      "../api/units:time_delta",
       "../rtc_base:checks",
       "../rtc_base:stringutils",
       "../test:fileutils",
diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index c3a41e4d11..6616243ac1 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -2910,7 +2910,7 @@ TEST_P(PeerConnectionIntegrationTest,
   // Some things use a time of "0" as a special value, so we need to start out
   // the fake clock at a nonzero time.
   // TODO(deadbeef): Fix this.
-  fake_clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
+  fake_clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
   virtual_socket_server()->set_delay_mean(5);  // 5 ms per hop.
   virtual_socket_server()->UpdateDelayDistribution();
 
@@ -3786,7 +3786,7 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndConnectionTimeWithTurnTurnPair) {
   // Some things use a time of "0" as a special value, so we need to start out
   // the fake clock at a nonzero time.
   // TODO(deadbeef): Fix this.
-  fake_clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
+  fake_clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
 
   static constexpr int media_hop_delay_ms = 50;
   static constexpr int signaling_trip_delay_ms = 500;
diff --git a/pc/rtcstatscollector_unittest.cc b/pc/rtcstatscollector_unittest.cc
index f27e953b56..08fa41537a 100644
--- a/pc/rtcstatscollector_unittest.cc
+++ b/pc/rtcstatscollector_unittest.cc
@@ -18,6 +18,7 @@
 #include "api/rtpparameters.h"
 #include "api/stats/rtcstats_objects.h"
 #include "api/stats/rtcstatsreport.h"
+#include "api/units/time_delta.h"
 #include "p2p/base/p2pconstants.h"
 #include "p2p/base/port.h"
 #include "pc/mediastream.h"
@@ -35,7 +36,6 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/ptr_util.h"
 #include "rtc_base/stringutils.h"
-#include "rtc_base/timedelta.h"
 #include "rtc_base/timeutils.h"
 
 using testing::AtLeast;
@@ -670,7 +670,7 @@ TEST_F(RTCStatsCollectorTest, CachedStatsReports) {
   rtc::scoped_refptr<const RTCStatsReport> c = stats_->GetStatsReport();
   EXPECT_NE(b.get(), c.get());
   // Invalidate cache by advancing time.
-  fake_clock_.AdvanceTime(rtc::TimeDelta::FromMilliseconds(51));
+  fake_clock_.AdvanceTime(TimeDelta::ms(51));
   rtc::scoped_refptr<const RTCStatsReport> d = stats_->GetStatsReport();
   EXPECT_TRUE(d);
   EXPECT_NE(c.get(), d.get());
@@ -681,7 +681,7 @@ TEST_F(RTCStatsCollectorTest, MultipleCallbacksWithInvalidatedCacheInBetween) {
   stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&a));
   stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&b));
   // Cache is invalidated after 50 ms.
-  fake_clock_.AdvanceTime(rtc::TimeDelta::FromMilliseconds(51));
+  fake_clock_.AdvanceTime(TimeDelta::ms(51));
   stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&c));
   EXPECT_TRUE_WAIT(a, kGetStatsReportTimeoutMs);
   EXPECT_TRUE_WAIT(b, kGetStatsReportTimeoutMs);