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/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h
index 0ac37e1..e574355 100644
--- a/webrtc/api/peerconnectioninterface.h
+++ b/webrtc/api/peerconnectioninterface.h
@@ -489,6 +489,17 @@
virtual IceConnectionState ice_connection_state() = 0;
virtual IceGatheringState ice_gathering_state() = 0;
+ // Starts RtcEventLog using existing file. Takes ownership of |file| and
+ // passes it on to Call, which will take the ownership. If the
+ // operation fails the file will be closed. The logging will stop
+ // automatically after 10 minutes have passed, or when the StopRtcEventLog
+ // function is called.
+ virtual bool StartRtcEventLog(rtc::PlatformFile file,
+ int64_t max_size_bytes) = 0;
+
+ // Stops logging the RtcEventLog.
+ virtual void StopRtcEventLog() = 0;
+
// Terminates all media and closes the transport.
virtual void Close() = 0;
@@ -655,25 +666,19 @@
// Stops logging the AEC dump.
virtual void StopAecDump() = 0;
- // Starts RtcEventLog using existing file. Takes ownership of |file| and
- // passes it on to VoiceEngine, which will take the ownership. If the
- // operation fails the file will be closed. The logging will stop
- // automatically after 10 minutes have passed, or when the StopRtcEventLog
- // function is called. A maximum filesize in bytes can be set, the logging
- // will be stopped before exceeding this limit. If max_size_bytes is set to a
- // value <= 0, no limit will be used.
- // This function as well as the StopRtcEventLog don't really belong on this
- // interface, this is a temporary solution until we move the logging object
- // from inside voice engine to webrtc::Call, which will happen when the VoE
- // restructuring effort is further along.
- // TODO(ivoc): Move this into being:
- // PeerConnection => MediaController => webrtc::Call.
+ // This function is deprecated and will be removed when Chrome is updated to
+ // use the equivalent function on PeerConnectionInterface.
+ // TODO(ivoc) Remove after Chrome is updated.
virtual bool StartRtcEventLog(rtc::PlatformFile file,
int64_t max_size_bytes) = 0;
- // Deprecated, use the version above.
+ // This function is deprecated and will be removed when Chrome is updated to
+ // use the equivalent function on PeerConnectionInterface.
+ // TODO(ivoc) Remove after Chrome is updated.
virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
- // Stops logging the RtcEventLog.
+ // This function is deprecated and will be removed when Chrome is updated to
+ // use the equivalent function on PeerConnectionInterface.
+ // TODO(ivoc) Remove after Chrome is updated.
virtual void StopRtcEventLog() = 0;
protected: