Create the SrtpTransportInterface.

webrtc/ortc/rtptransportcontrolleradapter.h

 /*
  *  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_RTPTRANSPORTCONTROLLERADAPTER_H_
 #define WEBRTC_ORTC_RTPTRANSPORTCONTROLLERADAPTER_H_
 
 #include <memory>
 #include <set>
 #include <string>
 #include <vector>
 
+#include "webrtc/api/ortc/ortcrtpreceiverinterface.h"
+#include "webrtc/api/ortc/ortcrtpsenderinterface.h"
+#include "webrtc/api/ortc/rtptransportcontrollerinterface.h"
+#include "webrtc/api/ortc/srtptransportinterface.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/call/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
-#include "webrtc/api/ortc/ortcrtpreceiverinterface.h"
-#include "webrtc/api/ortc/ortcrtpsenderinterface.h"
-#include "webrtc/api/ortc/rtptransportcontrollerinterface.h"
-#include "webrtc/api/ortc/rtptransportinterface.h"
+#include "webrtc/media/base/mediachannel.h"  // For MediaConfig.
 #include "webrtc/pc/channelmanager.h"
 #include "webrtc/pc/mediacontroller.h"
-#include "webrtc/media/base/mediachannel.h"  // For MediaConfig.
 
 namespace webrtc {
 
 class RtpTransportAdapter;
 class OrtcRtpSenderAdapter;
 class OrtcRtpReceiverAdapter;
 
 // Implementation of RtpTransportControllerInterface. Wraps a MediaController,
 // a VoiceChannel and VideoChannel, and maintains a list of dependent RTP
 // transports.
@@ skipping 33 matching lines @@
   std::vector<RtpTransportInterface*> GetTransports() const override;
 
   // These methods are used by OrtcFactory to create RtpTransports, RtpSenders
   // and RtpReceivers using this controller. Called "CreateProxied" because
   // these methods return proxies that will safely call methods on the correct
   // thread.
   RTCErrorOr<std::unique_ptr<RtpTransportInterface>> CreateProxiedRtpTransport(
       const RtcpParameters& rtcp_parameters,
       PacketTransportInterface* rtp,
       PacketTransportInterface* rtcp);
+
+  RTCErrorOr<std::unique_ptr<SrtpTransportInterface>>
+  CreateProxiedSrtpTransport(const RtcpParameters& rtcp_parameters,
+                             PacketTransportInterface* rtp,
+                             PacketTransportInterface* rtcp);
+
   // |transport_proxy| needs to be a proxy to a transport because the
   // application may call GetTransport() on the returned sender or receiver,
   // and expects it to return a thread-safe transport proxy.
   RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>> CreateProxiedRtpSender(
       cricket::MediaType kind,
       RtpTransportInterface* transport_proxy);
   RTCErrorOr<std::unique_ptr<OrtcRtpReceiverInterface>>
   CreateProxiedRtpReceiver(cricket::MediaType kind,
                            RtpTransportInterface* transport_proxy);
 
@@ skipping 69 matching lines @@
   // |description| is the matching description where existing SSRCs can be
   // found.
   //
   // This is a member function because it may need to generate SSRCs that don't
   // match existing ones, which is more than ToStreamParamsVec does.
   RTCErrorOr<cricket::StreamParamsVec> MakeSendStreamParamsVec(
       std::vector<RtpEncodingParameters> encodings,
       const std::string& cname,
       const cricket::MediaContentDescription& description) const;
 
+  // If the |rtp_transport| is a SrtpTransport, set the cryptos of the
+  // audio/video content descriptions.
+  RTCError MaybeSetCryptos(
+      RtpTransportInterface* rtp_transport,
+      cricket::MediaContentDescription* local_description,
+      cricket::MediaContentDescription* remote_description);
+
   rtc::Thread* signaling_thread_;
   rtc::Thread* worker_thread_;
   // |transport_proxies_| and |inner_audio_transport_|/|inner_audio_transport_|
   // are somewhat redundant, but the latter are only set when
   // RtpSenders/RtpReceivers are attached to the transport.
   std::vector<RtpTransportInterface*> transport_proxies_;
   RtpTransportInterface* inner_audio_transport_ = nullptr;
   RtpTransportInterface* inner_video_transport_ = nullptr;
   std::unique_ptr<MediaControllerInterface> media_controller_;
 
@@ skipping 10 matching lines @@
   bool have_video_sender_ = false;
   bool have_audio_receiver_ = false;
   bool have_video_receiver_ = false;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpTransportControllerAdapter);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_ORTC_RTPTRANSPORTCONTROLLERADAPTER_H_