Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(663)

Unified Diff: webrtc/call/call_perf_tests.cc

Issue 2353033005: Refactoring: move ownership of RtcEventLog from Call to PeerConnection (Closed)
Patch Set: Moved the constructor Created 4 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/call/call.cc ('k') | webrtc/call/call_unittest.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/call/call_perf_tests.cc
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index 4324d81e80d627c91f5b29be8cc16aa2266bf499..43d7aa5e068444d87758dfb4afb54df0503b82a6 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -19,6 +19,7 @@
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
#include "webrtc/config.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
@@ -165,9 +166,9 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
AudioState::Config send_audio_state_config;
send_audio_state_config.voice_engine = voice_engine;
- Call::Config sender_config;
+ Call::Config sender_config(&event_log_);
sender_config.audio_state = AudioState::Create(send_audio_state_config);
- Call::Config receiver_config;
+ Call::Config receiver_config(&event_log_);
receiver_config.audio_state = sender_config.audio_state;
CreateCalls(sender_config, receiver_config);
@@ -685,6 +686,7 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
Call::Config GetSenderCallConfig() override {
Call::Config config = EndToEndTest::GetSenderCallConfig();
+ config.event_log = &event_log_;
config.bitrate_config.start_bitrate_bps = kInitialBitrateKbps * 1000;
return config;
}
« no previous file with comments | « webrtc/call/call.cc ('k') | webrtc/call/call_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698