Index: webrtc/api/peerconnectioninterface.h |
diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h |
index 0ac37e1564e935584913bd0bebe8a1f720e2f6e1..e57435559e25c6a1254ad46b801afeb91e73f6ff 100644 |
--- a/webrtc/api/peerconnectioninterface.h |
+++ b/webrtc/api/peerconnectioninterface.h |
@@ -489,6 +489,17 @@ class PeerConnectionInterface : public rtc::RefCountInterface { |
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 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface { |
// 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: |