| /* | 
 |  *  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 PC_RTP_TRANSCEIVER_H_ | 
 | #define PC_RTP_TRANSCEIVER_H_ | 
 |  | 
 | #include <stddef.h> | 
 |  | 
 | #include <algorithm> | 
 | #include <functional> | 
 | #include <string> | 
 | #include <vector> | 
 |  | 
 | #include "absl/types/optional.h" | 
 | #include "api/array_view.h" | 
 | #include "api/media_types.h" | 
 | #include "api/proxy.h" | 
 | #include "api/rtc_error.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/rtp_transceiver_interface.h" | 
 | #include "api/scoped_refptr.h" | 
 | #include "api/task_queue/task_queue_base.h" | 
 | #include "pc/channel_interface.h" | 
 | #include "pc/channel_manager.h" | 
 | #include "pc/rtp_receiver.h" | 
 | #include "pc/rtp_sender.h" | 
 | #include "rtc_base/ref_counted_object.h" | 
 | #include "rtc_base/task_utils/pending_task_safety_flag.h" | 
 | #include "rtc_base/third_party/sigslot/sigslot.h" | 
 | #include "rtc_base/thread_annotations.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | // Implementation of the public RtpTransceiverInterface. | 
 | // | 
 | // The RtpTransceiverInterface is only intended to be used with a PeerConnection | 
 | // that enables Unified Plan SDP. Thus, the methods that only need to implement | 
 | // public API features and are not used internally can assume exactly one sender | 
 | // and receiver. | 
 | // | 
 | // Since the RtpTransceiver is used internally by PeerConnection for tracking | 
 | // RtpSenders, RtpReceivers, and BaseChannels, and PeerConnection needs to be | 
 | // backwards compatible with Plan B SDP, this implementation is more flexible | 
 | // than that required by the WebRTC specification. | 
 | // | 
 | // With Plan B SDP, an RtpTransceiver can have any number of senders and | 
 | // receivers which map to a=ssrc lines in the m= section. | 
 | // With Unified Plan SDP, an RtpTransceiver will have exactly one sender and one | 
 | // receiver which are encapsulated by the m= section. | 
 | // | 
 | // This class manages the RtpSenders, RtpReceivers, and BaseChannel associated | 
 | // with this m= section. Since the transceiver, senders, and receivers are | 
 | // reference counted and can be referenced from JavaScript (in Chromium), these | 
 | // objects must be ready to live for an arbitrary amount of time. The | 
 | // BaseChannel is not reference counted and is owned by the ChannelManager, so | 
 | // the PeerConnection must take care of creating/deleting the BaseChannel and | 
 | // setting the channel reference in the transceiver to null when it has been | 
 | // deleted. | 
 | // | 
 | // The RtpTransceiver is specialized to either audio or video according to the | 
 | // MediaType specified in the constructor. Audio RtpTransceivers will have | 
 | // AudioRtpSenders, AudioRtpReceivers, and a VoiceChannel. Video RtpTransceivers | 
 | // will have VideoRtpSenders, VideoRtpReceivers, and a VideoChannel. | 
 | class RtpTransceiver final | 
 |     : public rtc::RefCountedObject<RtpTransceiverInterface>, | 
 |       public sigslot::has_slots<> { | 
 |  public: | 
 |   // Construct a Plan B-style RtpTransceiver with no senders, receivers, or | 
 |   // channel set. | 
 |   // |media_type| specifies the type of RtpTransceiver (and, by transitivity, | 
 |   // the type of senders, receivers, and channel). Can either by audio or video. | 
 |   RtpTransceiver(cricket::MediaType media_type, | 
 |                  cricket::ChannelManager* channel_manager); | 
 |   // Construct a Unified Plan-style RtpTransceiver with the given sender and | 
 |   // receiver. The media type will be derived from the media types of the sender | 
 |   // and receiver. The sender and receiver should have the same media type. | 
 |   // |HeaderExtensionsToOffer| is used for initializing the return value of | 
 |   // HeaderExtensionsToOffer(). | 
 |   RtpTransceiver( | 
 |       rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender, | 
 |       rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>> | 
 |           receiver, | 
 |       cricket::ChannelManager* channel_manager, | 
 |       std::vector<RtpHeaderExtensionCapability> HeaderExtensionsToOffer, | 
 |       std::function<void()> on_negotiation_needed); | 
 |   ~RtpTransceiver() override; | 
 |  | 
 |   // Returns the Voice/VideoChannel set for this transceiver. May be null if | 
 |   // the transceiver is not in the currently set local/remote description. | 
 |   cricket::ChannelInterface* channel() const { return channel_; } | 
 |  | 
 |   // Sets the Voice/VideoChannel. The caller must pass in the correct channel | 
 |   // implementation based on the type of the transceiver. | 
 |   void SetChannel(cricket::ChannelInterface* channel); | 
 |  | 
 |   // Adds an RtpSender of the appropriate type to be owned by this transceiver. | 
 |   // Must not be null. | 
 |   void AddSender( | 
 |       rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender); | 
 |  | 
 |   // Removes the given RtpSender. Returns false if the sender is not owned by | 
 |   // this transceiver. | 
 |   bool RemoveSender(RtpSenderInterface* sender); | 
 |  | 
 |   // Returns a vector of the senders owned by this transceiver. | 
 |   std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>> | 
 |   senders() const { | 
 |     return senders_; | 
 |   } | 
 |  | 
 |   // Adds an RtpReceiver of the appropriate type to be owned by this | 
 |   // transceiver. Must not be null. | 
 |   void AddReceiver( | 
 |       rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>> | 
 |           receiver); | 
 |  | 
 |   // Removes the given RtpReceiver. Returns false if the sender is not owned by | 
 |   // this transceiver. | 
 |   bool RemoveReceiver(RtpReceiverInterface* receiver); | 
 |  | 
 |   // Returns a vector of the receivers owned by this transceiver. | 
 |   std::vector< | 
 |       rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>> | 
 |   receivers() const { | 
 |     return receivers_; | 
 |   } | 
 |  | 
 |   // Returns the backing object for the transceiver's Unified Plan sender. | 
 |   rtc::scoped_refptr<RtpSenderInternal> sender_internal() const; | 
 |  | 
 |   // Returns the backing object for the transceiver's Unified Plan receiver. | 
 |   rtc::scoped_refptr<RtpReceiverInternal> receiver_internal() const; | 
 |  | 
 |   // RtpTransceivers are not associated until they have a corresponding media | 
 |   // section set in SetLocalDescription or SetRemoteDescription. Therefore, | 
 |   // when setting a local offer we need a way to remember which transceiver was | 
 |   // used to create which media section in the offer. Storing the mline index | 
 |   // in CreateOffer is specified in JSEP to allow us to do that. | 
 |   absl::optional<size_t> mline_index() const { return mline_index_; } | 
 |   void set_mline_index(absl::optional<size_t> mline_index) { | 
 |     mline_index_ = mline_index; | 
 |   } | 
 |  | 
 |   // Sets the MID for this transceiver. If the MID is not null, then the | 
 |   // transceiver is considered "associated" with the media section that has the | 
 |   // same MID. | 
 |   void set_mid(const absl::optional<std::string>& mid) { mid_ = mid; } | 
 |  | 
 |   // Sets the intended direction for this transceiver. Intended to be used | 
 |   // internally over SetDirection since this does not trigger a negotiation | 
 |   // needed callback. | 
 |   void set_direction(RtpTransceiverDirection direction) { | 
 |     direction_ = direction; | 
 |   } | 
 |  | 
 |   // Sets the current direction for this transceiver as negotiated in an offer/ | 
 |   // answer exchange. The current direction is null before an answer with this | 
 |   // transceiver has been set. | 
 |   void set_current_direction(RtpTransceiverDirection direction); | 
 |  | 
 |   // Sets the fired direction for this transceiver. The fired direction is null | 
 |   // until SetRemoteDescription is called or an answer is set (either local or | 
 |   // remote). | 
 |   void set_fired_direction(RtpTransceiverDirection direction); | 
 |  | 
 |   // According to JSEP rules for SetRemoteDescription, RtpTransceivers can be | 
 |   // reused only if they were added by AddTrack. | 
 |   void set_created_by_addtrack(bool created_by_addtrack) { | 
 |     created_by_addtrack_ = created_by_addtrack; | 
 |   } | 
 |   // If AddTrack has been called then transceiver can't be removed during | 
 |   // rollback. | 
 |   void set_reused_for_addtrack(bool reused_for_addtrack) { | 
 |     reused_for_addtrack_ = reused_for_addtrack; | 
 |   } | 
 |  | 
 |   bool created_by_addtrack() const { return created_by_addtrack_; } | 
 |  | 
 |   bool reused_for_addtrack() const { return reused_for_addtrack_; } | 
 |  | 
 |   // Returns true if this transceiver has ever had the current direction set to | 
 |   // sendonly or sendrecv. | 
 |   bool has_ever_been_used_to_send() const { | 
 |     return has_ever_been_used_to_send_; | 
 |   } | 
 |  | 
 |   // Informs the transceiver that its owning | 
 |   // PeerConnection is closed. | 
 |   void SetPeerConnectionClosed(); | 
 |  | 
 |   // Executes the "stop the RTCRtpTransceiver" procedure from | 
 |   // the webrtc-pc specification, described under the stop() method. | 
 |   void StopTransceiverProcedure(); | 
 |  | 
 |   // Fired when the RtpTransceiver state changes such that negotiation is now | 
 |   // needed (e.g., in response to a direction change). | 
 |   //  sigslot::signal0<> SignalNegotiationNeeded; | 
 |  | 
 |   // RtpTransceiverInterface implementation. | 
 |   cricket::MediaType media_type() const override; | 
 |   absl::optional<std::string> mid() const override; | 
 |   rtc::scoped_refptr<RtpSenderInterface> sender() const override; | 
 |   rtc::scoped_refptr<RtpReceiverInterface> receiver() const override; | 
 |   bool stopped() const override; | 
 |   bool stopping() const override; | 
 |   RtpTransceiverDirection direction() const override; | 
 |   RTCError SetDirectionWithError( | 
 |       RtpTransceiverDirection new_direction) override; | 
 |   absl::optional<RtpTransceiverDirection> current_direction() const override; | 
 |   absl::optional<RtpTransceiverDirection> fired_direction() const override; | 
 |   RTCError StopStandard() override; | 
 |   void StopInternal() override; | 
 |   RTCError SetCodecPreferences( | 
 |       rtc::ArrayView<RtpCodecCapability> codecs) override; | 
 |   std::vector<RtpCodecCapability> codec_preferences() const override { | 
 |     return codec_preferences_; | 
 |   } | 
 |   std::vector<RtpHeaderExtensionCapability> HeaderExtensionsToOffer() | 
 |       const override; | 
 |   std::vector<RtpHeaderExtensionCapability> HeaderExtensionsNegotiated() | 
 |       const override; | 
 |   RTCError SetOfferedRtpHeaderExtensions( | 
 |       rtc::ArrayView<const RtpHeaderExtensionCapability> | 
 |           header_extensions_to_offer) override; | 
 |  | 
 |   // Called on the signaling thread when the local or remote content description | 
 |   // is updated. Used to update the negotiated header extensions. | 
 |   // TODO(tommi): The implementation of this method is currently very simple and | 
 |   // only used for updating the negotiated headers. However, we're planning to | 
 |   // move all the updates done on the channel from the transceiver into this | 
 |   // method. This will happen with the ownership of the channel object being | 
 |   // moved into the transceiver. | 
 |   void OnNegotiationUpdate(SdpType sdp_type, | 
 |                            const cricket::MediaContentDescription* content); | 
 |  | 
 |  private: | 
 |   void OnFirstPacketReceived(); | 
 |   void StopSendingAndReceiving(); | 
 |  | 
 |   // Enforce that this object is created, used and destroyed on one thread. | 
 |   TaskQueueBase* const thread_; | 
 |   const bool unified_plan_; | 
 |   const cricket::MediaType media_type_; | 
 |   rtc::scoped_refptr<PendingTaskSafetyFlag> signaling_thread_safety_; | 
 |   std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>> | 
 |       senders_; | 
 |   std::vector< | 
 |       rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>> | 
 |       receivers_; | 
 |  | 
 |   bool stopped_ RTC_GUARDED_BY(thread_) = false; | 
 |   bool stopping_ RTC_GUARDED_BY(thread_) = false; | 
 |   bool is_pc_closed_ = false; | 
 |   RtpTransceiverDirection direction_ = RtpTransceiverDirection::kInactive; | 
 |   absl::optional<RtpTransceiverDirection> current_direction_; | 
 |   absl::optional<RtpTransceiverDirection> fired_direction_; | 
 |   absl::optional<std::string> mid_; | 
 |   absl::optional<size_t> mline_index_; | 
 |   bool created_by_addtrack_ = false; | 
 |   bool reused_for_addtrack_ = false; | 
 |   bool has_ever_been_used_to_send_ = false; | 
 |  | 
 |   cricket::ChannelInterface* channel_ = nullptr; | 
 |   cricket::ChannelManager* channel_manager_ = nullptr; | 
 |   std::vector<RtpCodecCapability> codec_preferences_; | 
 |   std::vector<RtpHeaderExtensionCapability> header_extensions_to_offer_; | 
 |  | 
 |   // |negotiated_header_extensions_| is read and written to on the signaling | 
 |   // thread from the SdpOfferAnswerHandler class (e.g. | 
 |   // PushdownMediaDescription(). | 
 |   cricket::RtpHeaderExtensions negotiated_header_extensions_ | 
 |       RTC_GUARDED_BY(thread_); | 
 |  | 
 |   const std::function<void()> on_negotiation_needed_; | 
 | }; | 
 |  | 
 | BEGIN_PRIMARY_PROXY_MAP(RtpTransceiver) | 
 |  | 
 | PROXY_PRIMARY_THREAD_DESTRUCTOR() | 
 | BYPASS_PROXY_CONSTMETHOD0(cricket::MediaType, media_type) | 
 | PROXY_CONSTMETHOD0(absl::optional<std::string>, mid) | 
 | PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpSenderInterface>, sender) | 
 | PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpReceiverInterface>, receiver) | 
 | PROXY_CONSTMETHOD0(bool, stopped) | 
 | PROXY_CONSTMETHOD0(bool, stopping) | 
 | PROXY_CONSTMETHOD0(RtpTransceiverDirection, direction) | 
 | PROXY_METHOD1(webrtc::RTCError, SetDirectionWithError, RtpTransceiverDirection) | 
 | PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, current_direction) | 
 | PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, fired_direction) | 
 | PROXY_METHOD0(webrtc::RTCError, StopStandard) | 
 | PROXY_METHOD0(void, StopInternal) | 
 | PROXY_METHOD1(webrtc::RTCError, | 
 |               SetCodecPreferences, | 
 |               rtc::ArrayView<RtpCodecCapability>) | 
 | PROXY_CONSTMETHOD0(std::vector<RtpCodecCapability>, codec_preferences) | 
 | PROXY_CONSTMETHOD0(std::vector<RtpHeaderExtensionCapability>, | 
 |                    HeaderExtensionsToOffer) | 
 | PROXY_CONSTMETHOD0(std::vector<RtpHeaderExtensionCapability>, | 
 |                    HeaderExtensionsNegotiated) | 
 | PROXY_METHOD1(webrtc::RTCError, | 
 |               SetOfferedRtpHeaderExtensions, | 
 |               rtc::ArrayView<const RtpHeaderExtensionCapability>) | 
 | END_PROXY_MAP() | 
 |  | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // PC_RTP_TRANSCEIVER_H_ |