commit d5bdda328e913715ef1af2667883eb9351c7c3e2
author pbos <pbos@webrtc.org> Mon Sep 28 16:59:31 2015
committer Commit bot <commit-bot@chromium.org> Mon Sep 28 16:59:36 2015
tree 09e05456b07af3ca47469a29d13ca8ca025f5df2
parent 7a17ca656cb015d66a661c344997967d73dcef71

Unify Transport and newapi::Transport interfaces.

BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1369263002

Cr-Original-Commit-Position: refs/heads/master@{#10096}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 2d566686a23fe93ada58f1c38a0d4b9a0d68556e

test/call_test.cc

diff --git a/test/call_test.cc b/test/call_test.cc
index 19e292f..0986df5 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -90,7 +90,7 @@
 }
 
 void CallTest::CreateSendConfig(size_t num_streams,
-                                newapi::Transport* send_transport) {
+                                Transport* send_transport) {
   assert(num_streams <= kNumSsrcs);
   send_config_ = VideoSendStream::Config(send_transport);
   send_config_.encoder_settings.encoder = &fake_encoder_;
@@ -106,7 +106,7 @@
 }
 
 void CallTest::CreateMatchingReceiveConfigs(
-    newapi::Transport* rtcp_send_transport) {
+    Transport* rtcp_send_transport) {
   assert(!send_config_.rtp.ssrcs.empty());
   assert(receive_configs_.empty());
   assert(allocated_decoders_.empty());