| 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
|
|
|