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Unified Diff: webrtc/api/peerconnectioninterface.h

Issue 1748403002: Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Another rebase and accompanying changes. Created 4 years, 6 months ago
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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:
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