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

Issue 1426443007: Revert of Adding the ability to create an RtpSender without a track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 2 months ago
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Index: talk/app/webrtc/mediastreamprovider.h
diff --git a/talk/app/webrtc/mediastreamprovider.h b/talk/app/webrtc/mediastreamprovider.h
index a78b55a68c0a73e95cae68b9ab923c0b5a10200b..1c62daf9f16aa33d0874b524970e1b41dbcbff28 100644
--- a/talk/app/webrtc/mediastreamprovider.h
+++ b/talk/app/webrtc/mediastreamprovider.h
@@ -50,8 +50,8 @@
// RtpSenders/Receivers to get to the BaseChannels. These interfaces should be
// refactored away eventually, as the classes converge.
-// This interface is called by AudioRtpSender/Receivers to change the settings
-// of an audio track connected to certain PeerConnection.
+// This interface is called by AudioTrackHandler classes in mediastreamhandler.h
+// to change the settings of an audio track connected to certain PeerConnection.
class AudioProviderInterface {
public:
// Enable/disable the audio playout of a remote audio track with |ssrc|.
@@ -71,8 +71,9 @@
virtual ~AudioProviderInterface() {}
};
-// This interface is called by VideoRtpSender/Receivers to change the settings
-// of a video track connected to a certain PeerConnection.
+// This interface is called by VideoTrackHandler classes in mediastreamhandler.h
+// to change the settings of a video track connected to a certain
+// PeerConnection.
class VideoProviderInterface {
public:
virtual bool SetCaptureDevice(uint32_t ssrc,
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