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Unified Diff: talk/app/webrtc/peerconnection.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/peerconnection.h
diff --git a/talk/app/webrtc/peerconnection.h b/talk/app/webrtc/peerconnection.h
index c47f90372e3e9d0586cf045e90731b5516989a22..ffe2e409e169b5f033b12aa69f0aecd132bcb668 100644
--- a/talk/app/webrtc/peerconnection.h
+++ b/talk/app/webrtc/peerconnection.h
@@ -93,6 +93,9 @@ class PeerConnection : public PeerConnectionInterface,
rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
AudioTrackInterface* track) override;
+ rtc::scoped_refptr<RtpSenderInterface> CreateSender(
+ const std::string& kind) override;
+
std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
const override;
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
@@ -189,12 +192,6 @@ class PeerConnection : public PeerConnectionInterface,
AudioTrackInterface* audio_track);
void DestroyVideoReceiver(MediaStreamInterface* stream,
VideoTrackInterface* video_track);
- void CreateAudioSender(MediaStreamInterface* stream,
- AudioTrackInterface* audio_track,
- uint32_t ssrc);
- void CreateVideoSender(MediaStreamInterface* stream,
- VideoTrackInterface* video_track,
- uint32_t ssrc);
void DestroyAudioSender(MediaStreamInterface* stream,
AudioTrackInterface* audio_track,
uint32_t ssrc);
@@ -330,6 +327,8 @@ class PeerConnection : public PeerConnectionInterface,
void OnDataChannelOpenMessage(const std::string& label,
const InternalDataChannelInit& config);
+ RtpSenderInterface* FindSenderById(const std::string& id);
+
std::vector<rtc::scoped_refptr<RtpSenderInterface>>::iterator
FindSenderForTrack(MediaStreamTrackInterface* track);
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>::iterator

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