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;
 
-// 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(
+// TODO(zhihuang): Remove this declaration when the WebRtcSession tests don't
+// need it.
+void ExtractSharedMediaSessionOptions(
     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.
-  virtual bool GetOptionsForOffer(
+  void GetOptionsForOffer(
       const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
       cricket::MediaSessionOptions* session_options);
 
   // Returns a MediaSessionOptions struct with options decided by
   // |constraints|, the local MediaStreams and DataChannels.
-  // 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);
+  void GetOptionsForAnswer(const RTCOfferAnswerOptions& options,
+                           cricket::MediaSessionOptions* session_options);
 
-  void InitializeOptionsForAnswer(
-      cricket::MediaSessionOptions* session_options);
-
-  // Helper function for options processing.
-  // Deprecated.
-  virtual void FinishOptionsForAnswer(
+  // 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,
+      rtc::Optional<size_t>* audio_index,
+      rtc::Optional<size_t>* video_index,
+      rtc::Optional<size_t>* data_index,
       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_