commit ef9daee934107d9ff911a0db3b4582fd6a62d4ec
author Sebastian Jansson <srte@webrtc.org> Thu Feb 22 14:49:02 2018 +0100
committer Commit Bot <commit-bot@chromium.org> Thu Feb 22 17:32:25 2018 +0000
tree 915e6b35c18b0ddeb5d1bbd39dfc923219ed9570
parent 89c79383e45f9c46951f74e8971fb9fedfe0137c

Using mock transport controller in audio unit tests.

Using a mock of rtp transport controller send in audio send stream unit
tests. This reduces the dependencies and makes the tests more focused
on testing the functionality of audio send stream itself.

Bug: webrtc:8415
Change-Id: Ia8d9cf47d93decc74b10ca75a6771f39df658dc2
Reviewed-on: https://webrtc-review.googlesource.com/56600
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22161}

audio/BUILD.gn

diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 59302ea..618ec84 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -125,9 +125,9 @@
       ":audio_end_to_end_test",
       "../api:mock_audio_mixer",
       "../call:mock_call_interfaces",
+      "../call:mock_rtp_interfaces",
       "../call:rtp_interfaces",
       "../call:rtp_receiver",
-      "../call:rtp_sender",
       "../common_audio",
       "../logging:mocks",
       "../modules:module_api",