180 lines
8.3 KiB
C
180 lines
8.3 KiB
C
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/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_RTP_TRANSCEIVER_INTERFACE_H_
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#define API_RTP_TRANSCEIVER_INTERFACE_H_
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#include <string>
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#include <vector>
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#include "absl/base/attributes.h"
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/media_types.h"
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#include "api/rtp_parameters.h"
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#include "api/rtp_receiver_interface.h"
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#include "api/rtp_sender_interface.h"
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#include "api/rtp_transceiver_direction.h"
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#include "api/scoped_refptr.h"
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#include "rtc_base/ref_count.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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// Structure for initializing an RtpTransceiver in a call to
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// PeerConnectionInterface::AddTransceiver.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
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struct RTC_EXPORT RtpTransceiverInit final {
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RtpTransceiverInit();
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RtpTransceiverInit(const RtpTransceiverInit&);
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~RtpTransceiverInit();
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// Direction of the RtpTransceiver. See RtpTransceiverInterface::direction().
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RtpTransceiverDirection direction = RtpTransceiverDirection::kSendRecv;
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// The added RtpTransceiver will be added to these streams.
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std::vector<std::string> stream_ids;
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// TODO(bugs.webrtc.org/7600): Not implemented.
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std::vector<RtpEncodingParameters> send_encodings;
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};
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// The RtpTransceiverInterface maps to the RTCRtpTransceiver defined by the
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// WebRTC specification. A transceiver represents a combination of an RtpSender
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// and an RtpReceiver than share a common mid. As defined in JSEP, an
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// RtpTransceiver is said to be associated with a media description if its mid
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// property is non-null; otherwise, it is said to be disassociated.
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// JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24
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//
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// Note that RtpTransceivers are only supported when using PeerConnection with
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// Unified Plan SDP.
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//
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// This class is thread-safe.
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//
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// WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
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class RTC_EXPORT RtpTransceiverInterface : public rtc::RefCountInterface {
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public:
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// Media type of the transceiver. Any sender(s)/receiver(s) will have this
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// type as well.
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virtual cricket::MediaType media_type() const = 0;
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// The mid attribute is the mid negotiated and present in the local and
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// remote descriptions. Before negotiation is complete, the mid value may be
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// null. After rollbacks, the value may change from a non-null value to null.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
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virtual absl::optional<std::string> mid() const = 0;
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// The sender attribute exposes the RtpSender corresponding to the RTP media
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// that may be sent with the transceiver's mid. The sender is always present,
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// regardless of the direction of media.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-sender
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virtual rtc::scoped_refptr<RtpSenderInterface> sender() const = 0;
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// The receiver attribute exposes the RtpReceiver corresponding to the RTP
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// media that may be received with the transceiver's mid. The receiver is
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// always present, regardless of the direction of media.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-receiver
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virtual rtc::scoped_refptr<RtpReceiverInterface> receiver() const = 0;
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// The stopped attribute indicates that the sender of this transceiver will no
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// longer send, and that the receiver will no longer receive. It is true if
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// either stop has been called or if setting the local or remote description
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// has caused the RtpTransceiver to be stopped.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
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virtual bool stopped() const = 0;
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// The stopping attribute indicates that the user has indicated that the
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// sender of this transceiver will stop sending, and that the receiver will
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// no longer receive. It is always true if stopped() is true.
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// If stopping() is true and stopped() is false, it means that the
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// transceiver's stop() method has been called, but the negotiation with
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// the other end for shutting down the transceiver is not yet done.
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// https://w3c.github.io/webrtc-pc/#dfn-stopping-0
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// TODO(hta): Remove default implementation.
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virtual bool stopping() const;
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// The direction attribute indicates the preferred direction of this
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// transceiver, which will be used in calls to CreateOffer and CreateAnswer.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
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virtual RtpTransceiverDirection direction() const = 0;
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// Sets the preferred direction of this transceiver. An update of
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// directionality does not take effect immediately. Instead, future calls to
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// CreateOffer and CreateAnswer mark the corresponding media descriptions as
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// sendrecv, sendonly, recvonly, or inactive.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
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// TODO(hta): Deprecate SetDirection without error and rename
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// SetDirectionWithError to SetDirection, remove default implementations.
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ABSL_DEPRECATED("Use SetDirectionWithError instead")
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virtual void SetDirection(RtpTransceiverDirection new_direction);
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virtual RTCError SetDirectionWithError(RtpTransceiverDirection new_direction);
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// The current_direction attribute indicates the current direction negotiated
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// for this transceiver. If this transceiver has never been represented in an
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// offer/answer exchange, or if the transceiver is stopped, the value is null.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
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virtual absl::optional<RtpTransceiverDirection> current_direction() const = 0;
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// An internal slot designating for which direction the relevant
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// PeerConnection events have been fired. This is to ensure that events like
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// OnAddTrack only get fired once even if the same session description is
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// applied again.
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// Exposed in the public interface for use by Chromium.
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virtual absl::optional<RtpTransceiverDirection> fired_direction() const;
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// Initiates a stop of the transceiver.
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// The stop is complete when stopped() returns true.
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// A stopped transceiver can be reused for a different track.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
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// TODO(hta): Rename to Stop() when users of the non-standard Stop() are
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// updated.
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virtual RTCError StopStandard();
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// Stops a transceiver immediately, without waiting for signalling.
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// This is an internal function, and is exposed for historical reasons.
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// https://w3c.github.io/webrtc-pc/#dfn-stop-the-rtcrtptransceiver
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virtual void StopInternal();
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ABSL_DEPRECATED("Use StopStandard instead") virtual void Stop();
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// The SetCodecPreferences method overrides the default codec preferences used
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// by WebRTC for this transceiver.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-setcodecpreferences
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virtual RTCError SetCodecPreferences(
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rtc::ArrayView<RtpCodecCapability> codecs);
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virtual std::vector<RtpCodecCapability> codec_preferences() const;
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// Readonly attribute which contains the set of header extensions that was set
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// with SetOfferedRtpHeaderExtensions, or a default set if it has not been
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// called.
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// https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface
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virtual std::vector<RtpHeaderExtensionCapability> HeaderExtensionsToOffer()
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const;
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// Readonly attribute which is either empty if negotation has not yet
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// happened, or a vector of the negotiated header extensions.
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// https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface
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virtual std::vector<RtpHeaderExtensionCapability> HeaderExtensionsNegotiated()
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const;
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// The SetOfferedRtpHeaderExtensions method modifies the next SDP negotiation
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// so that it negotiates use of header extensions which are not kStopped.
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// https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface
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virtual webrtc::RTCError SetOfferedRtpHeaderExtensions(
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rtc::ArrayView<const RtpHeaderExtensionCapability>
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header_extensions_to_offer);
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protected:
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~RtpTransceiverInterface() override = default;
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};
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} // namespace webrtc
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#endif // API_RTP_TRANSCEIVER_INTERFACE_H_
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