Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1028)

Unified Diff: talk/app/webrtc/peerconnectioninterface.h

Issue 1413983004: Reland of Adding the ability to create an RtpSender without a track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing merge issue. Created 5 years, 1 month ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « talk/app/webrtc/peerconnection_unittest.cc ('k') | talk/app/webrtc/peerconnectioninterface_unittest.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: talk/app/webrtc/peerconnectioninterface.h
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index b7814e583033ac4f741c8d1b245ed14e5c12c143..723d643784dab7c83ad826c21133af17f095ef0a 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -333,6 +333,12 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
AudioTrackInterface* track) = 0;
// TODO(deadbeef): Make these pure virtual once all subclasses implement them.
+ // |kind| must be "audio" or "video".
+ virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
+ const std::string& kind) {
+ return rtc::scoped_refptr<RtpSenderInterface>();
+ }
+
virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
const {
return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
« no previous file with comments | « talk/app/webrtc/peerconnection_unittest.cc ('k') | talk/app/webrtc/peerconnectioninterface_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698