Wire up RTP keep-alive in ortc api.

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/call/call.h"
+#include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/media/base/mediachannel.h"  // For MediaConfig.
 #include "webrtc/pc/channelmanager.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/sigslot.h"
 #include "webrtc/rtc_base/thread.h"
 
 namespace webrtc {
 
 class RtpTransportAdapter;
@@ skipping 36 matching lines @@
   ~RtpTransportControllerAdapter() override;
 
   // RtpTransportControllerInterface implementation.
   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,
+      const RtpTransportParameters& rtcp_parameters,
       PacketTransportInterface* rtp,
       PacketTransportInterface* rtcp);
 
   RTCErrorOr<std::unique_ptr<SrtpTransportInterface>>
-  CreateProxiedSrtpTransport(const RtcpParameters& rtcp_parameters,
+  CreateProxiedSrtpTransport(const RtpTransportParameters& 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);
 
   // Methods used internally by other "adapter" classes.
   rtc::Thread* signaling_thread() const { return signaling_thread_; }
   rtc::Thread* worker_thread() const { return worker_thread_; }
 
-  RTCError SetRtcpParameters(const RtcpParameters& parameters,
-                             RtpTransportInterface* inner_transport);
+  // |parameters.keepalive| will be set for ALL RTP transports in the call.
+  RTCError SetRtpTransportParameters(const RtpTransportParameters& parameters,
+                                     RtpTransportInterface* inner_transport);
+  void SetRtpTransportParameters_w(const RtpTransportParameters& parameters);
 
   cricket::VoiceChannel* voice_channel() { return voice_channel_; }
   cricket::VideoChannel* video_channel() { return video_channel_; }
 
   // |primary_ssrc| out parameter is filled with either
   // |parameters.encodings[0].ssrc|, or a generated SSRC if that's left unset.
   RTCError ValidateAndApplyAudioSenderParameters(
       const RtpParameters& parameters,
       uint32_t* primary_ssrc);
   RTCError ValidateAndApplyVideoSenderParameters(
@@ skipping 71 matching lines @@
 
   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;
   const cricket::MediaConfig media_config_;
+  RtpKeepAliveConfig keepalive_;
   cricket::ChannelManager* channel_manager_;
   webrtc::RtcEventLog* event_log_;
   std::unique_ptr<Call> call_;
+  webrtc::RtpTransportControllerSend* call_send_rtp_transport_controller_;
 
   // BaseChannel takes content descriptions as input, so we store them here
   // such that they can be updated when a new RtpSenderAdapter/
   // RtpReceiverAdapter attaches itself.
   cricket::AudioContentDescription local_audio_description_;
   cricket::AudioContentDescription remote_audio_description_;
   cricket::VideoContentDescription local_video_description_;
   cricket::VideoContentDescription remote_video_description_;
   cricket::VoiceChannel* voice_channel_ = nullptr;
   cricket::VideoChannel* video_channel_ = nullptr;
   bool have_audio_sender_ = false;
   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_