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

Issue 1413713003: Adding the ability to create an RtpSender without a track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Adding some unit tests for new methods on the sender. Created 5 years, 2 months ago
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Index: talk/app/webrtc/peerconnectioninterface.h
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index d3d0d9e66710d8a97cdf3b24529c2f3ac8c59085..b285212541aba1acea143da40c71aebffccee97e 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -337,6 +337,12 @@ 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".
pthatcher1 2015/10/20 17:42:49 I think we could use cricket::MediaType here and s
Taylor Brandstetter 2015/10/21 00:22:08 Done.
+ 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>>();

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