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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/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/ref_count.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;
// 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 RTC_EXPORT 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 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;
// Readonly attribute which contains the set of header extensions that was set
// with SetOfferedRtpHeaderExtensions, or a default set if it has not been
// called.
// https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface
virtual std::vector<RtpHeaderExtensionCapability> HeaderExtensionsToOffer()
const = 0;
// Readonly attribute which is either empty 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> HeaderExtensionsNegotiated()
const = 0;
// The SetOfferedRtpHeaderExtensions 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 SetOfferedRtpHeaderExtensions(
rtc::ArrayView<const RtpHeaderExtensionCapability>
header_extensions_to_offer) = 0;
protected:
~RtpTransceiverInterface() override = default;
};
} // namespace webrtc
#endif // API_RTP_TRANSCEIVER_INTERFACE_H_
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