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Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.

Browse Source 
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
This commit is contained in:
wu@webrtc.org
2013-12-12 22:40:39 +00:00
parent 451745ec05
commit a129b6cd13
101 changed files with 490 additions and 290 deletions
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8
talk/app/webrtc/peerconnectioninterface.h
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@@ -393,11 +393,9 @@ class PeerConnectionFactoryInterface : public talk_base::RefCountInterface {
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 @@ class PeerConnectionFactoryInterface : public talk_base::RefCountInterface {
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.