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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"

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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}
This commit is contained in:
ivoc
2016-07-04 07:06:55 -07:00
committed by Commit bot
parent 77ad394fa6
commit 14d5dbe5b3
61 changed files with 428 additions and 359 deletions
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39
webrtc/api/peerconnectioninterface.h
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@ -493,6 +493,21 @@ 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.
// 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 @@ 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: