Index: webrtc/media/engine/webrtcvoiceengine.h |
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h |
index d60953439eb0494d98da8007e95089f34e7a1d27..64e0f5b1850d9860b23c69751c2f2883ebc7dfec 100644 |
--- a/webrtc/media/engine/webrtcvoiceengine.h |
+++ b/webrtc/media/engine/webrtcvoiceengine.h |
@@ -109,14 +109,6 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback { |
// 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 |
// ignored. This allows us to selectively turn on and off different options |