| Index: talk/media/webrtc/webrtcvoiceengine.h
|
| diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
|
| index a70840565b433c6188152eecd41719d416ee2cf2..1cf05e71a26a8bfefca3010aee1b66cc7bc860a7 100644
|
| --- a/talk/media/webrtc/webrtcvoiceengine.h
|
| +++ b/talk/media/webrtc/webrtcvoiceengine.h
|
| @@ -108,6 +108,9 @@ class WebRtcVoiceEngine
|
| // Starts AEC dump using existing file.
|
| bool StartAecDump(rtc::PlatformFile file);
|
|
|
| + // Stops AEC dump.
|
| + void StopAecDump();
|
| +
|
| // Starts recording an RtcEventLog using an existing file until 10 minutes
|
| // pass or the StopRtcEventLog function is called.
|
| bool StartRtcEventLog(rtc::PlatformFile file);
|
| @@ -139,7 +142,6 @@ class WebRtcVoiceEngine
|
| bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
|
|
|
| void StartAecDump(const std::string& filename);
|
| - void StopAecDump();
|
| int CreateVoEChannel();
|
|
|
| static const int kDefaultLogSeverity = rtc::LS_WARNING;
|
|
|