Delete MediaController class, move Call ownership to PeerConnection.

webrtc/pc/webrtcsession.h

 /*
  *  Copyright 2012 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_PC_WEBRTCSESSION_H_
 #define WEBRTC_PC_WEBRTCSESSION_H_
 
 #include <memory>
 #include <set>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/statstypes.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/p2p/base/candidate.h"
 #include "webrtc/p2p/base/transportcontroller.h"
 #include "webrtc/pc/datachannel.h"
-#include "webrtc/pc/mediacontroller.h"
 #include "webrtc/pc/mediasession.h"
 
 #ifdef HAVE_QUIC
 #include "webrtc/pc/quicdatatransport.h"
 #endif  // HAVE_QUIC
 
 namespace cricket {
 
 class ChannelManager;
 class RtpDataChannel;
 class SctpTransportInternal;
 class SctpTransportInternalFactory;
 class StatsReport;
 class VideoChannel;
 class VoiceChannel;
 
 #ifdef HAVE_QUIC
 class QuicTransportChannel;
 #endif  // HAVE_QUIC
 
 }  // namespace cricket
 
 namespace webrtc {
 
 class IceRestartAnswerLatch;
 class JsepIceCandidate;
 class MediaStreamSignaling;
+class RtcEventLog;
 class WebRtcSessionDescriptionFactory;
 
 extern const char kBundleWithoutRtcpMux[];
 extern const char kCreateChannelFailed[];
 extern const char kInvalidCandidates[];
 extern const char kInvalidSdp[];
 extern const char kMlineMismatch[];
 extern const char kPushDownTDFailed[];
 extern const char kSdpWithoutDtlsFingerprint[];
 extern const char kSdpWithoutSdesCrypto[];
@@ skipping 84 matching lines @@
   };
 
   enum Error {
     ERROR_NONE = 0,       // no error
     ERROR_CONTENT = 1,    // channel errors in SetLocalContent/SetRemoteContent
     ERROR_TRANSPORT = 2,  // transport error of some kind
   };
 
   // |sctp_factory| may be null, in which case SCTP is treated as unsupported.
   WebRtcSession(
-      webrtc::MediaControllerInterface* media_controller,
+      cricket::ChannelManager* channel_manager,
+      const cricket::MediaConfig& media_config,
+      RtcEventLog* event_log,
       rtc::Thread* network_thread,
       rtc::Thread* worker_thread,
       rtc::Thread* signaling_thread,
       cricket::PortAllocator* port_allocator,
       std::unique_ptr<cricket::TransportController> transport_controller,
       std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory);
   virtual ~WebRtcSession();
 
   // These are const to allow them to be called from const methods.
   rtc::Thread* network_thread() const { return network_thread_; }
@@ skipping 182 matching lines @@
   // Called when a valid data channel OPEN message is received.
   // std::string represents the data channel label.
   sigslot::signal2<const std::string&, const InternalDataChannelInit&>
       SignalDataChannelOpenMessage;
 #ifdef HAVE_QUIC
   QuicDataTransport* quic_data_transport() {
     return quic_data_transport_.get();
   }
 #endif  // HAVE_QUIC
 
+  // To let tests override the Call object we use.
+ protected:
+  virtual Call* call() { return call_.get(); };

[nisse-webrtc, 2017/04/27 13:46:34]

This is a bit ugly. It's needed *only* for the test

WebRtcSessionTest.TestPacketOptionsAndOnPacketSent

In Call, it looks like packet_id is used for SendDelayStats::OnSentPacket. Can
we hook in something there instead?

An alternative might be to pass Call to the WebRtcSession constructor, then
ownership would have to be moved one step further, to PeerConnection.

[Taylor Brandstetter, 2017/04/30 07:21:29]

I think it would make more sense to pass a Call into the WebRtcSession
constructor (optionally). It may only be one test, but this is a level of
interaction we probably should be testing more of.

+
  private:
   // Indicates the type of SessionDescription in a call to SetLocalDescription
   // and SetRemoteDescription.
   enum Action {
     kOffer,
     kPrAnswer,
     kAnswer,
   };
 
   // Non-const versions of local_description()/remote_description(), for use
@@ skipping 164 matching lines @@
 
   void OnSentPacket_w(const rtc::SentPacket& sent_packet);
 
   const std::string GetTransportName(const std::string& content_name);
 
   void DestroyRtcpTransport_n(const std::string& transport_name);
   void DestroyVideoChannel();
   void DestroyVoiceChannel();
   void DestroyDataChannel();
 
+  void Init_w();
+  void Close_w();
+
   rtc::Thread* const network_thread_;
   rtc::Thread* const worker_thread_;
   rtc::Thread* const signaling_thread_;
 
   State state_ = STATE_INIT;
   Error error_ = ERROR_NONE;
   std::string error_desc_;
 
   const std::string sid_;
   bool initial_offerer_ = false;
 
   const std::unique_ptr<cricket::TransportController> transport_controller_;
   const std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory_;
-  MediaControllerInterface* media_controller_;
+  const cricket::MediaConfig media_config_;
+  RtcEventLog* event_log_;
+  std::unique_ptr<Call> call_;
   std::unique_ptr<cricket::VoiceChannel> voice_channel_;
   std::unique_ptr<cricket::VideoChannel> video_channel_;
   // |rtp_data_channel_| is used if in RTP data channel mode, |sctp_transport_|
   // when using SCTP.
   std::unique_ptr<cricket::RtpDataChannel> rtp_data_channel_;
 
   std::unique_ptr<cricket::SctpTransportInternal> sctp_transport_;
   // |sctp_transport_name_| keeps track of what DTLS transport the SCTP
   // transport is using (which can change due to bundling).
   rtc::Optional<std::string> sctp_transport_name_;
@@ skipping 60 matching lines @@
 
 #ifdef HAVE_QUIC
   std::unique_ptr<QuicDataTransport> quic_data_transport_;
 #endif  // HAVE_QUIC
 
   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_PC_WEBRTCSESSION_H_