Add RtpTransceiverInterface and implementing class
Introduces the public API interface corresponding to the
standardized RtpTransceiver object in the WebRTC spec.
https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
The RtpTransceiver will be the internal representation for both
Plan B and Unified Plan SDP, but the public API interface will
only support Unified Plan (existing users should continue to use
GetSenders/GetReceivers, which will still be supported).
Bug: webrtc:7600
Change-Id: I417ffda683209ba9a9b4cbd274f91ca8295779a7
Reviewed-on: https://webrtc-review.googlesource.com/21460
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20659}diff --git a/api/rtptransceiverinterface.h b/api/rtptransceiverinterface.h
new file mode 100644
index 0000000..3eb246a
--- /dev/null
+++ b/api/rtptransceiverinterface.h
@@ -0,0 +1,107 @@
+/*
+ * 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 "api/optional.h"
+#include "api/rtpreceiverinterface.h"
+#include "api/rtpsenderinterface.h"
+#include "rtc_base/refcount.h"
+
+namespace webrtc {
+
+enum class RtpTransceiverDirection {
+ kSendRecv,
+ kSendOnly,
+ kRecvOnly,
+ kInactive
+};
+
+// 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:
+ // 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_