Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
The breaking tests in Chromium have been temporarily disabled, they will be fixed and reenabled soon.
Original CLs: https://codereview.webrtc.org/1748403002/, https://codereview.webrtc.org/2107253002/ and https://codereview.webrtc.org/2106103003/.
TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org,tkchin@webrtc.org
BUG=webrtc:4741, webrtc:5603, chromium:609749
Review-Url: https://codereview.webrtc.org/2110113003
Cr-Commit-Position: refs/heads/master@{#13379}
diff --git a/webrtc/modules/bitrate_controller/include/bitrate_controller.h b/webrtc/modules/bitrate_controller/include/bitrate_controller.h
index 4165f06..f06314c 100644
--- a/webrtc/modules/bitrate_controller/include/bitrate_controller.h
+++ b/webrtc/modules/bitrate_controller/include/bitrate_controller.h
@@ -55,8 +55,11 @@
// TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
// Remove this method once other other projects does not use it.
static BitrateController* CreateBitrateController(Clock* clock,
- BitrateObserver* observer);
- static BitrateController* CreateBitrateController(Clock* clock);
+ BitrateObserver* observer,
+ RtcEventLog* event_log);
+
+ static BitrateController* CreateBitrateController(Clock* clock,
+ RtcEventLog* event_log);
virtual ~BitrateController() {}
@@ -76,8 +79,6 @@
virtual void UpdateDelayBasedEstimate(uint32_t bitrate_bps) = 0;
- virtual void SetEventLog(RtcEventLog* event_log) = 0;
-
// Gets the available payload bandwidth in bits per second. Note that
// this bandwidth excludes packet headers.
virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;