Index: webrtc/ortc/ortcrtpsenderadapter.h |
diff --git a/webrtc/ortc/ortcrtpsenderadapter.h b/webrtc/ortc/ortcrtpsenderadapter.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..9b60f1530097c72899af00f5e8bf357ef9aaa964 |
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+++ b/webrtc/ortc/ortcrtpsenderadapter.h |
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+/* |
+ * Copyright 2017 The WebRTC project authors. All Rights Reserved. |
+ * |
+ * Use of this source code is governed by a BSD-style license |
+ * that can be found in the LICENSE file in the root of the source |
+ * tree. An additional intellectual property rights grant can be found |
+ * in the file PATENTS. All contributing project authors may |
+ * be found in the AUTHORS file in the root of the source tree. |
+ */ |
+ |
+#ifndef WEBRTC_ORTC_ORTCRTPSENDERADAPTER_H_ |
+#define WEBRTC_ORTC_ORTCRTPSENDERADAPTER_H_ |
+ |
+#include <memory> |
+ |
+#include "webrtc/api/ortc/ortcrtpsenderinterface.h" |
+#include "webrtc/api/rtcerror.h" |
+#include "webrtc/api/rtpparameters.h" |
+#include "webrtc/base/constructormagic.h" |
+#include "webrtc/base/sigslot.h" |
+#include "webrtc/ortc/rtptransportcontrolleradapter.h" |
+#include "webrtc/pc/rtpsender.h" |
+ |
+namespace webrtc { |
+ |
+// Implementation of OrtcRtpSenderInterface that works with RtpTransportAdapter, |
+// and wraps a VideoRtpSender/AudioRtpSender that's normally used with the |
+// PeerConnection. |
+// |
+// TODO(deadbeef): When BaseChannel is split apart into separate |
+// "RtpSender"/"RtpTransceiver"/"RtpSender"/"RtpReceiver" objects, this adapter |
+// object can be removed. |
+class OrtcRtpSenderAdapter : public OrtcRtpSenderInterface { |
+ public: |
+ // Wraps |wrapped_sender| in a proxy that will safely call methods on the |
+ // correct thread. |
+ static std::unique_ptr<OrtcRtpSenderInterface> CreateProxy( |
+ std::unique_ptr<OrtcRtpSenderAdapter> wrapped_sender); |
+ |
+ // Should only be called by RtpTransportControllerAdapter. |
+ OrtcRtpSenderAdapter(cricket::MediaType kind, |
+ RtpTransportInterface* transport, |
+ RtpTransportControllerAdapter* rtp_transport_controller); |
+ ~OrtcRtpSenderAdapter() override; |
+ |
+ // OrtcRtpSenderInterface implementation. |
+ RTCError SetTrack(MediaStreamTrackInterface* track) override; |
+ rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const override; |
+ |
+ RTCError SetTransport(RtpTransportInterface* transport) override; |
+ RtpTransportInterface* GetTransport() const override; |
+ |
+ RTCError Send(const RtpParameters& parameters) override; |
+ RtpParameters GetParameters() const override; |
+ |
+ cricket::MediaType GetKind() const override; |
+ |
+ // Used so that the RtpTransportControllerAdapter knows when it can |
+ // deallocate resources allocated for this object. |
+ sigslot::signal0<> SignalDestroyed; |
+ |
+ private: |
+ void CreateInternalSender(); |
+ |
+ cricket::MediaType kind_; |
+ RtpTransportInterface* transport_; |
+ RtpTransportControllerAdapter* rtp_transport_controller_; |
+ // Scoped refptr due to ref-counted interface, but we should be the only |
+ // reference holder. |
+ rtc::scoped_refptr<RtpSenderInternal> internal_sender_; |
+ rtc::scoped_refptr<MediaStreamTrackInterface> track_; |
+ RtpParameters last_applied_parameters_; |
+ |
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(OrtcRtpSenderAdapter); |
+}; |
+ |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_ORTC_ORTCRTPSENDERADAPTER_H_ |