Index: talk/media/webrtc/webrtcvoiceengine.h |
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h |
index cd792bcb435843fe2320cbc7bd3d42fa658220a2..290b52b635541a13e0c4cf2180a8748bdfc847c9 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); |
@@ -142,7 +145,6 @@ 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 CreateVoiceChannel(VoEWrapper* voe); |
static const int kDefaultLogSeverity = rtc::LS_WARNING; |