Move RtcEventLog object from inside VoiceEngine to Call.
In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
BUG=webrtc:4741,webrtc:5603,chromium:609749
R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1748403002 .
Cr-Commit-Position: refs/heads/master@{#13321}
diff --git a/webrtc/modules/bitrate_controller/bitrate_controller_impl.h b/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
index 91f0902..114e1a6 100644
--- a/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
+++ b/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
@@ -30,7 +30,9 @@
public:
// TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
// |observer| is left for project that is not yet updated.
- BitrateControllerImpl(Clock* clock, BitrateObserver* observer);
+ BitrateControllerImpl(Clock* clock,
+ BitrateObserver* observer,
+ RtcEventLog* event_log);
virtual ~BitrateControllerImpl() {}
bool AvailableBandwidth(uint32_t* bandwidth) const override;
@@ -54,8 +56,6 @@
void SetReservedBitrate(uint32_t reserved_bitrate_bps) override;
- void SetEventLog(RtcEventLog* event_log) override;
-
// Returns true if the parameters have changed since the last call.
bool GetNetworkParameters(uint32_t* bitrate,
uint8_t* fraction_loss,
@@ -86,6 +86,7 @@
Clock* const clock_;
BitrateObserver* const observer_;
int64_t last_bitrate_update_ms_;
+ RtcEventLog* const event_log_;
rtc::CriticalSection critsect_;
SendSideBandwidthEstimation bandwidth_estimation_ GUARDED_BY(critsect_);