Non-constraint interfaces for all constrainable interfaces

webrtc/api/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 @@
 
 namespace webrtc {
 
 class MediaStreamObserver;
 class RemoteMediaStreamFactory;
 
 // 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 ConvertRtcOptionsForOffer(
+bool ExtractMediaSessionOptions(
     const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
     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);
 
 // Parses the URLs for each server in |servers| to build |stun_servers| and
 // |turn_servers|.
 bool ParseIceServers(const PeerConnectionInterface::IceServers& servers,
                      cricket::ServerAddresses* stun_servers,
                      std::vector<cricket::RelayServerConfig>* turn_servers);
 
 // 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);
 
   bool Initialize(
+      const cricket::MediaConfig& media_config,
       const PeerConnectionInterface::RTCConfiguration& configuration,
-      const MediaConstraintsInterface* constraints,
       rtc::scoped_ptr<cricket::PortAllocator> allocator,
       rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
       PeerConnectionObserver* observer);
 
   rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
   rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
   bool AddStream(MediaStreamInterface* local_stream) override;
   void RemoveStream(MediaStreamInterface* local_stream) override;
 
   rtc::scoped_refptr<RtpSenderInterface> AddTrack(
@@ skipping 26 matching lines @@
 
   // TODO(bemasc): Remove ice_state() when callers are removed.
   IceState ice_state() override;
   IceConnectionState ice_connection_state() override;
   IceGatheringState ice_gathering_state() override;
 
   const SessionDescriptionInterface* local_description() const override;
   const SessionDescriptionInterface* remote_description() const override;
 
   // JSEP01
+  // Deprecated, use version without constraints.
   void CreateOffer(CreateSessionDescriptionObserver* observer,
                    const MediaConstraintsInterface* constraints) override;
   void CreateOffer(CreateSessionDescriptionObserver* observer,
                    const RTCOfferAnswerOptions& options) override;
+  // Deprecated, use version without constraints.
   void CreateAnswer(CreateSessionDescriptionObserver* observer,
                     const MediaConstraintsInterface* constraints) override;
+  void CreateAnswer(CreateSessionDescriptionObserver* observer,
+                    const RTCOfferAnswerOptions& options) override;
   void SetLocalDescription(SetSessionDescriptionObserver* observer,
                            SessionDescriptionInterface* desc) override;
   void SetRemoteDescription(SetSessionDescriptionObserver* observer,
                             SessionDescriptionInterface* desc) override;
   bool SetConfiguration(
       const PeerConnectionInterface::RTCConfiguration& config) override;
   bool AddIceCandidate(const IceCandidateInterface* candidate) override;
 
   void RegisterUMAObserver(UMAObserver* observer) override;
 
@@ skipping 79 matching lines @@
   }
 
   // Returns a MediaSessionOptions struct with options decided by |options|,
   // the local MediaStreams and DataChannels.
   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.
+  // 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);
+
+  // 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.
   // |media_type| is the type of the |streams| and can be either audio or video.
@@ skipping 149 matching lines @@
   // because its destruction fires signals (such as VoiceChannelDestroyed)
   // which will trigger some final actions in PeerConnection...
   rtc::scoped_ptr<WebRtcSession> session_;
   // ... But stats_ depends on session_ so it should be destroyed even earlier.
   rtc::scoped_ptr<StatsCollector> stats_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_PEERCONNECTION_H_