commit 8e6602fadeb26db9d66af4a6dea8e675fc133ab6
author Sebastian Jansson <srte@webrtc.org> Fri Jul 13 08:43:20 2018
committer Commit Bot <commit-bot@chromium.org> Fri Jul 13 10:27:37 2018
tree f6f6162f35399e4eece7643a0bc08e0bea02c408
parent bbbe4e1a15edbbfe99bb13f69593414865f720f2

Separates send and receive event log in CallTest.

This makes it possible to use them in VideoQualityTest and prepares for
allowing saving logs in other tests as well.

Bug: webrtc:9510
Change-Id: I3b1cc187d88e4f17745414433c2f96efd836a302
Reviewed-on: https://webrtc-review.googlesource.com/88561
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23964}

call/call_perf_tests.cc

diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index dd70e32..40f47fe 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -179,12 +179,12 @@
     send_audio_state_config.audio_processing =
         AudioProcessingBuilder().Create();
     send_audio_state_config.audio_device_module = fake_audio_device;
-    Call::Config sender_config(event_log_.get());
+    Call::Config sender_config(send_event_log_.get());
 
     auto audio_state = AudioState::Create(send_audio_state_config);
     fake_audio_device->RegisterAudioCallback(audio_state->audio_transport());
     sender_config.audio_state = audio_state;
-    Call::Config receiver_config(event_log_.get());
+    Call::Config receiver_config(recv_event_log_.get());
     receiver_config.audio_state = audio_state;
     CreateCalls(sender_config, receiver_config);