Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(56)

Unified Diff: webrtc/media/engine/fakewebrtccall.h

Issue 1748403002: Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: More fixes for bots. Created 4 years, 9 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
Index: webrtc/media/engine/fakewebrtccall.h
diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h
index b1e20ce4c05ae4c4ea96767a83c07bcabdc5f7aa..1acb8428ed9197d5de7b1d7056fef21598ef3283 100644
--- a/webrtc/media/engine/fakewebrtccall.h
+++ b/webrtc/media/engine/fakewebrtccall.h
@@ -239,6 +239,10 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
webrtc::NetworkState state) override;
void OnSentPacket(const rtc::SentPacket& sent_packet) override;
+ bool StartEventLog(rtc::PlatformFile log_file,
+ int64_t max_size_bytes) override;
+ void StopEventLog() override;
+
webrtc::Call::Config config_;
webrtc::NetworkState audio_network_state_;
webrtc::NetworkState video_network_state_;

Powered by Google App Engine
This is Rietveld 408576698