Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
R=mallinath@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index a127dad..01f1e1c 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -393,11 +393,9 @@
class Options {
public:
Options() :
- enable_aec_dump(false),
disable_encryption(false),
disable_sctp_data_channels(false) {
}
- bool enable_aec_dump;
bool disable_encryption;
bool disable_sctp_data_channels;
};
@@ -442,6 +440,12 @@
CreateAudioTrack(const std::string& label,
AudioSourceInterface* source) = 0;
+ // Starts AEC dump using existing file. Takes ownership of |file| and passes
+ // it on to VoiceEngine (via other objects) immediately, which will take
+ // the ownerhip.
+ // TODO(grunell): Remove when Chromium has started to use AEC in each source.
+ virtual bool StartAecDump(FILE* file) = 0;
+
protected:
// Dtor and ctor protected as objects shouldn't be created or deleted via
// this interface.