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}

audio/BUILD.gn

diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 8981dd3..75a62c2 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -128,6 +128,7 @@
       ":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",