text: /*
   *  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 API_RTPTRANSCEIVERINTERFACE_H_
  #define API_RTPTRANSCEIVERINTERFACE_H_
  
  #include <string>
  #include <vector>
  
  #include "api/array_view.h"
  #include "api/optional.h"
  #include "api/rtpreceiverinterface.h"
  #include "api/rtpsenderinterface.h"
  #include "rtc_base/refcount.h"
  
  namespace webrtc {
  
  // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverdirection
  enum class RtpTransceiverDirection {
    kSendRecv,
    kSendOnly,
    kRecvOnly,
    kInactive
  };
  
  // This is provided as a debugging aid. The format of the output is unspecified.
  std::ostream& operator<<(std::ostream& os, RtpTransceiverDirection direction);
  
  // Structure for initializing an RtpTransceiver in a call to
  // PeerConnectionInterface::AddTransceiver.
  // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
  struct RtpTransceiverInit final {
    // Direction of the RtpTransceiver. See RtpTransceiverInterface::direction().
    RtpTransceiverDirection direction = RtpTransceiverDirection::kSendRecv;
  
    // The added RtpTransceiver will be added to these streams.
    std::vector<std::string> stream_ids;
  
    // TODO(bugs.webrtc.org/7600): Not implemented.
    std::vector<RtpEncodingParameters> send_encodings;
  };
  
  // The RtpTransceiverInterface maps to the RTCRtpTransceiver defined by the
  // WebRTC specification. A transceiver represents a combination of an RtpSender
  // and an RtpReceiver than share a common mid. As defined in JSEP, an
  // RtpTransceiver is said to be associated with a media description if its mid
  // property is non-null; otherwise, it is said to be disassociated.
  // JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24
  //
  // Note that RtpTransceivers are only supported when using PeerConnection with
  // Unified Plan SDP.
  //
  // This class is thread-safe.
  //
  // WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
  // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
  class RtpTransceiverInterface : public rtc::RefCountInterface {
   public:
    // Media type of the transceiver. Any sender(s)/receiver(s) will have this
    // type as well.
    virtual cricket::MediaType media_type() const = 0;
  
    // The mid attribute is the mid negotiated and present in the local and
    // remote descriptions. Before negotiation is complete, the mid value may be
    // null. After rollbacks, the value may change from a non-null value to null.
    // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
    virtual rtc::Optional<std::string> mid() const = 0;
  
    // The sender attribute exposes the RtpSender corresponding to the RTP media
    // that may be sent with the transceiver's mid. The sender is always present,
    // regardless of the direction of media.
    // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-sender
    virtual rtc::scoped_refptr<RtpSenderInterface> sender() const = 0;
  
    // The receiver attribute exposes the RtpReceiver corresponding to the RTP
    // media that may be received with the transceiver's mid. The receiver is
    // always present, regardless of the direction of media.
    // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-receiver
    virtual rtc::scoped_refptr<RtpReceiverInterface> receiver() const = 0;
  
    // The stopped attribute indicates that the sender of this transceiver will no
    // longer send, and that the receiver will no longer receive. It is true if
    // either stop has been called or if setting the local or remote description
    // has caused the RtpTransceiver to be stopped.
    // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
    virtual bool stopped() const = 0;
  
    // The direction attribute indicates the preferred direction of this
    // transceiver, which will be used in calls to CreateOffer and CreateAnswer.
    // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
    virtual RtpTransceiverDirection direction() const = 0;
  
    // Sets the preferred direction of this transceiver. An update of
    // directionality does not take effect immediately. Instead, future calls to
    // CreateOffer and CreateAnswer mark the corresponding media descriptions as
    // sendrecv, sendonly, recvonly, or inactive.
    // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
    virtual void SetDirection(RtpTransceiverDirection new_direction) = 0;
  
    // The current_direction attribute indicates the current direction negotiated
    // for this transceiver. If this transceiver has never been represented in an
    // offer/answer exchange, or if the transceiver is stopped, the value is null.
    // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
    virtual rtc::Optional<RtpTransceiverDirection> current_direction() const = 0;
  
    // The Stop method irreversibly stops the RtpTransceiver. The sender of this
    // transceiver will no longer send, the receiver will no longer receive.
    // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
    virtual void Stop() = 0;
  
    // The SetCodecPreferences method overrides the default codec preferences used
    // by WebRTC for this transceiver.
    // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-setcodecpreferences
    // TODO(steveanton): Not implemented.
    virtual void SetCodecPreferences(
        rtc::ArrayView<RtpCodecCapability> codecs) = 0;
  
   protected:
    virtual ~RtpTransceiverInterface() = default;
  };
  
  }  // namespace webrtc
  
  #endif  // API_RTPTRANSCEIVERINTERFACE_H_
  