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TMessagesProj/jni/voip/webrtc/api/rtp_transceiver_interface.h

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+/*
+ *  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_RTP_TRANSCEIVER_INTERFACE_H_
+#define API_RTP_TRANSCEIVER_INTERFACE_H_
+
+#include <string>
+#include <vector>
+
+#include "absl/base/attributes.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/media_types.h"
+#include "api/ref_count.h"
+#include "api/rtp_parameters.h"
+#include "api/rtp_receiver_interface.h"
+#include "api/rtp_sender_interface.h"
+#include "api/rtp_transceiver_direction.h"
+#include "api/scoped_refptr.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Structure for initializing an RtpTransceiver in a call to
+// PeerConnectionInterface::AddTransceiver.
+// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
+struct RTC_EXPORT RtpTransceiverInit final {
+  RtpTransceiverInit();
+  RtpTransceiverInit(const RtpTransceiverInit&);
+  ~RtpTransceiverInit();
+  // 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;
+
+  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 RTC_EXPORT RtpTransceiverInterface : public webrtc::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 absl::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 stopping attribute indicates that the user has indicated that the
+  // sender of this transceiver will stop sending, and that the receiver will
+  // no longer receive. It is always true if stopped() is true.
+  // If stopping() is true and stopped() is false, it means that the
+  // transceiver's stop() method has been called, but the negotiation with
+  // the other end for shutting down the transceiver is not yet done.
+  // https://w3c.github.io/webrtc-pc/#dfn-stopping-0
+  virtual bool stopping() 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
+  // TODO(hta): Deprecate SetDirection without error and rename
+  // SetDirectionWithError to SetDirection, remove default implementations.
+  ABSL_DEPRECATED("Use SetDirectionWithError instead")
+  virtual void SetDirection(RtpTransceiverDirection new_direction);
+  virtual RTCError SetDirectionWithError(RtpTransceiverDirection new_direction);
+
+  // 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 absl::optional<RtpTransceiverDirection> current_direction() const = 0;
+
+  // An internal slot designating for which direction the relevant
+  // PeerConnection events have been fired. This is to ensure that events like
+  // OnAddTrack only get fired once even if the same session description is
+  // applied again.
+  // Exposed in the public interface for use by Chromium.
+  virtual absl::optional<RtpTransceiverDirection> fired_direction() const;
+
+  // Initiates a stop of the transceiver.
+  // The stop is complete when stopped() returns true.
+  // A stopped transceiver can be reused for a different track.
+  // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
+  // TODO(hta): Rename to Stop() when users of the non-standard Stop() are
+  // updated.
+  virtual RTCError StopStandard();
+
+  // Stops a transceiver immediately, without waiting for signalling.
+  // This is an internal function, and is exposed for historical reasons.
+  // https://w3c.github.io/webrtc-pc/#dfn-stop-the-rtcrtptransceiver
+  virtual void StopInternal();
+  ABSL_DEPRECATED("Use StopStandard instead") virtual void Stop();
+
+  // The SetCodecPreferences method overrides the default codec preferences used
+  // by WebRTC for this transceiver.
+  // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-setcodecpreferences
+  virtual RTCError SetCodecPreferences(
+      rtc::ArrayView<RtpCodecCapability> codecs) = 0;
+  virtual std::vector<RtpCodecCapability> codec_preferences() const = 0;
+
+  // Returns the set of header extensions that was set
+  // with SetHeaderExtensionsToNegotiate, or a default set if it has not been
+  // called.
+  // https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface
+  virtual std::vector<RtpHeaderExtensionCapability>
+  GetHeaderExtensionsToNegotiate() const = 0;
+
+  // Returns either the empty set if negotation has not yet
+  // happened, or a vector of the negotiated header extensions.
+  // https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface
+  virtual std::vector<RtpHeaderExtensionCapability>
+  GetNegotiatedHeaderExtensions() const = 0;
+
+  // The SetHeaderExtensionsToNegotiate method modifies the next SDP negotiation
+  // so that it negotiates use of header extensions which are not kStopped.
+  // https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface
+  virtual webrtc::RTCError SetHeaderExtensionsToNegotiate(
+      rtc::ArrayView<const RtpHeaderExtensionCapability> header_extensions) = 0;
+
+ protected:
+  ~RtpTransceiverInterface() override = default;
+};
+
+}  // namespace webrtc
+
+#endif  // API_RTP_TRANSCEIVER_INTERFACE_H_