commit 14d5dbe5b35c560e978dab0c02c48cb16d28273d
author ivoc <ivoc@webrtc.org> Mon Jul 04 07:06:55 2016 -0700
committer Commit bot <commit-bot@chromium.org> Mon Jul 04 14:07:03 2016 +0000
tree aa73a0a6a99a86ab13489ea0358f7235c7bf1ae9
parent 77ad394fa6680aa9fb36d0bcac330dd84b29dfc2

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}

webrtc/api/peerconnectioninterface.h

diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h
index e28a27e..39c4856 100644
--- a/webrtc/api/peerconnectioninterface.h
+++ b/webrtc/api/peerconnectioninterface.h
@@ -493,6 +493,21 @@
   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.
+  // TODO(ivoc): Make this pure virtual when Chrome is updated.
+  virtual bool StartRtcEventLog(rtc::PlatformFile file,
+                                int64_t max_size_bytes) {
+    return false;
+  }
+
+  // Stops logging the RtcEventLog.
+  // TODO(ivoc): Make this pure virtual when Chrome is updated.
+  virtual void StopRtcEventLog() {}
+
   // Terminates all media and closes the transport.
   virtual void Close() = 0;
 
@@ -659,25 +674,19 @@
   // 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: