Index: webrtc/media/engine/webrtcvoiceengine.h |
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h |
index 16da9ef29ed0d6be5a678a7aa7f9c2eb8ff23d41..52983fad2dd8bae45e2de0f287e84d04a285c813 100644 |
--- a/webrtc/media/engine/webrtcvoiceengine.h |
+++ b/webrtc/media/engine/webrtcvoiceengine.h |
@@ -108,6 +108,14 @@ |
// Stops AEC dump. |
void StopAecDump(); |
+ |
+ // Starts recording an RtcEventLog using an existing file until the log file |
+ // reaches the maximum filesize or the StopRtcEventLog function is called. |
+ // If the value of max_size_bytes is <= 0, no limit is used. |
+ bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes); |
+ |
+ // Stops recording the RtcEventLog. |
+ void StopRtcEventLog(); |
private: |
// Every option that is "set" will be applied. Every option not "set" will be |