| /* |
| * 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/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. |
| explicit RtpTransceiver(cricket::MediaType media_type); |
| // 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; |
| |
| private: |
| void OnFirstPacketReceived(cricket::ChannelInterface* channel); |
| void StopSendingAndReceiving(); |
| |
| // Enforce that this object is created, used and destroyed on one thread. |
| const TaskQueueBase* thread_; |
| const bool unified_plan_; |
| const cricket::MediaType media_type_; |
| std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>> |
| senders_; |
| std::vector< |
| rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>> |
| receivers_; |
| |
| bool stopped_ = 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_; |
| 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_ |