Revert of Adding support for Unified Plan offer/answer negotiation.

webrtc/pc/peerconnection.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.
  */
 
@@ skipping 14 matching lines @@
 #include "webrtc/pc/statscollector.h"
 #include "webrtc/pc/streamcollection.h"
 #include "webrtc/pc/webrtcsession.h"
 
 namespace webrtc {
 
 class MediaStreamObserver;
 class VideoRtpReceiver;
 class RtcEventLog;
 
-// TODO(zhihuang): Remove this declaration when the WebRtcSession tests don't
-// need it.
-void ExtractSharedMediaSessionOptions(
+// Populates |session_options| from |rtc_options|, and returns true if options
+// are valid.
+// |session_options|->transport_options map entries must exist in order for
+// them to be populated from |rtc_options|.
+bool ExtractMediaSessionOptions(
     const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
+    bool is_offer,
     cricket::MediaSessionOptions* session_options);
 
+// Populates |session_options| from |constraints|, and returns true if all
+// mandatory constraints are satisfied.
+// Assumes that |session_options|->transport_options map entries exist.
+// Will also set defaults if corresponding constraints are not present:
+// recv_audio=true, recv_video=true, bundle_enabled=true.
+// Other fields will be left with existing values.
+//
+// Deprecated. Will be removed once callers that use constraints are gone.
+// TODO(hta): Remove when callers are gone.
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5617
+bool ParseConstraintsForAnswer(const MediaConstraintsInterface* constraints,
+                               cricket::MediaSessionOptions* session_options);
+
 // PeerConnection implements the PeerConnectionInterface interface.
 // It uses WebRtcSession to implement the PeerConnection functionality.
 class PeerConnection : public PeerConnectionInterface,
                        public IceObserver,
                        public rtc::MessageHandler,
                        public sigslot::has_slots<> {
  public:
   explicit PeerConnection(PeerConnectionFactory* factory,
                           std::unique_ptr<RtcEventLog> event_log,
                           std::unique_ptr<Call> call);
@@ skipping 170 matching lines @@
   void PostCreateSessionDescriptionFailure(
       CreateSessionDescriptionObserver* observer,
       const std::string& error);
 
   bool IsClosed() const {
     return signaling_state_ == PeerConnectionInterface::kClosed;
   }
 
   // Returns a MediaSessionOptions struct with options decided by |options|,
   // the local MediaStreams and DataChannels.
-  void GetOptionsForOffer(
+  virtual bool GetOptionsForOffer(
       const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
       cricket::MediaSessionOptions* session_options);
 
   // Returns a MediaSessionOptions struct with options decided by
   // |constraints|, the local MediaStreams and DataChannels.
-  void GetOptionsForAnswer(const RTCOfferAnswerOptions& options,
-                           cricket::MediaSessionOptions* session_options);
+  // Deprecated, use version without constraints.
+  virtual bool GetOptionsForAnswer(
+      const MediaConstraintsInterface* constraints,
+      cricket::MediaSessionOptions* session_options);
+  virtual bool GetOptionsForAnswer(
+      const RTCOfferAnswerOptions& options,
+      cricket::MediaSessionOptions* session_options);
 
-  // Generates MediaDescriptionOptions for the |session_opts| based on existing
-  // local description or remote description.
-  void GenerateMediaDescriptionOptions(
-      const SessionDescriptionInterface* session_desc,
-      cricket::RtpTransceiverDirection audio_direction,
-      cricket::RtpTransceiverDirection video_direction,
-      int* audio_index,
-      int* video_index,
-      int* data_index,
+  void InitializeOptionsForAnswer(
+      cricket::MediaSessionOptions* session_options);
+
+  // Helper function for options processing.
+  // Deprecated.
+  virtual void FinishOptionsForAnswer(
       cricket::MediaSessionOptions* session_options);
 
   // Remove all local and remote tracks of type |media_type|.
   // Called when a media type is rejected (m-line set to port 0).
   void RemoveTracks(cricket::MediaType media_type);
 
   // Makes sure a MediaStreamTrack is created for each StreamParam in |streams|,
   // and existing MediaStreamTracks are removed if there is no corresponding
   // StreamParam. If |default_track_needed| is true, a default MediaStreamTrack
   // is created if it doesn't exist; if false, it's removed if it exists.
@@ skipping 77 matching lines @@
   void OnVoiceChannelDestroyed();
   void OnVideoChannelCreated();
   void OnVideoChannelDestroyed();
   void OnDataChannelCreated();
   void OnDataChannelDestroyed();
   // Called when the cricket::DataChannel receives a message indicating that a
   // webrtc::DataChannel should be opened.
   void OnDataChannelOpenMessage(const std::string& label,
                                 const InternalDataChannelInit& config);
 
-  bool HasRtpSender(cricket::MediaType type) const;
   RtpSenderInternal* FindSenderById(const std::string& id);
 
   std::vector<rtc::scoped_refptr<
       RtpSenderProxyWithInternal<RtpSenderInternal>>>::iterator
   FindSenderForTrack(MediaStreamTrackInterface* track);
   std::vector<rtc::scoped_refptr<
       RtpReceiverProxyWithInternal<RtpReceiverInternal>>>::iterator
   FindReceiverForTrack(const std::string& track_id);
 
   TrackInfos* GetRemoteTracks(cricket::MediaType media_type);
@@ skipping 83 matching lines @@
   std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
       senders_;
   std::vector<
       rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
       receivers_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_PC_PEERCONNECTION_H_