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| author | Sebastian Jansson <srte@webrtc.org> | 2018-05-08 14:52:22 +0200 |
|---|---|---|
| committer | Commit Bot <commit-bot@chromium.org> | 2018-05-08 13:22:53 +0000 |
| commit | 5f83cf0c6d842352d8e017321f0223aefebb1bef (patch) | |
| tree | 3a82358adf0e282e0c0de4c58ab85c22a0d1409e /audio | |
| parent | 5b2b692079e65be33998e6b8e7e7b7efce3cf9bf (diff) | |
| download | webrtc-5f83cf0c6d842352d8e017321f0223aefebb1bef.tar.gz | |
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}
Diffstat (limited to 'audio')
| -rw-r--r-- | audio/BUILD.gn | 1 | ||||
| -rw-r--r-- | audio/audio_send_stream_unittest.cc | 5 | ||||
| -rw-r--r-- | audio/time_interval_unittest.cc | 4 |
3 files changed, 5 insertions, 5 deletions
diff --git a/audio/BUILD.gn b/audio/BUILD.gn index 8981dd3353..75a62c2f75 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -128,6 +128,7 @@ if (rtc_include_tests) { ":audio_end_to_end_test", "../api:mock_audio_mixer", "../api/audio:audio_frame_api", + "../api/units:time_delta", "../call:mock_call_interfaces", "../call:mock_rtp_interfaces", "../call:rtp_interfaces", diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc index 8fb7e7e197..d39912000c 100644 --- a/audio/audio_send_stream_unittest.cc +++ b/audio/audio_send_stream_unittest.cc @@ -12,6 +12,7 @@ #include <utility> #include <vector> +#include "api/units/time_delta.h" #include "audio/audio_send_stream.h" #include "audio/audio_state.h" #include "audio/conversion.h" @@ -28,7 +29,6 @@ #include "rtc_base/fakeclock.h" #include "rtc_base/ptr_util.h" #include "rtc_base/task_queue.h" -#include "rtc_base/timedelta.h" #include "test/gtest.h" #include "test/mock_audio_encoder.h" #include "test/mock_audio_encoder_factory.h" @@ -545,8 +545,7 @@ TEST(AudioSendStreamTest, UpdateLifetime) { EXPECT_CALL(mock_transport, SendRtp(_, _, _)).Times(2); const PacketOptions options; registered_transport->SendRtp(nullptr, 0, options); - fake_clock.AdvanceTime( - rtc::TimeDelta::FromMilliseconds(kTimeBetweenSendRtpCallsMs)); + fake_clock.AdvanceTime(TimeDelta::ms(kTimeBetweenSendRtpCallsMs)); registered_transport->SendRtp(nullptr, 0, options); } EXPECT_TRUE(!helper.active_lifetime()->Empty()); diff --git a/audio/time_interval_unittest.cc b/audio/time_interval_unittest.cc index 7f8b44ecec..deff6e363d 100644 --- a/audio/time_interval_unittest.cc +++ b/audio/time_interval_unittest.cc @@ -9,8 +9,8 @@ */ #include "audio/time_interval.h" +#include "api/units/time_delta.h" #include "rtc_base/fakeclock.h" -#include "rtc_base/timedelta.h" #include "test/gtest.h" namespace webrtc { @@ -19,7 +19,7 @@ TEST(TimeIntervalTest, TimeInMs) { rtc::ScopedFakeClock fake_clock; TimeInterval interval; interval.Extend(); - fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(100)); + fake_clock.AdvanceTime(TimeDelta::ms(100)); interval.Extend(); EXPECT_EQ(interval.Length(), 100); } |