// Copyright 2020 The Chromium Authors. All rights reserved. // Use of this source code is governed by a BSD-style license that can be // found in the LICENSE file. #ifndef CAST_STREAMING_SENDER_H_ #define CAST_STREAMING_SENDER_H_ #include #include #include #include #include "absl/types/span.h" #include "cast/streaming/compound_rtcp_parser.h" #include "cast/streaming/constants.h" #include "cast/streaming/frame_crypto.h" #include "cast/streaming/frame_id.h" #include "cast/streaming/rtp_defines.h" #include "cast/streaming/rtp_packetizer.h" #include "cast/streaming/rtp_time.h" #include "cast/streaming/sender_packet_router.h" #include "cast/streaming/sender_report_builder.h" #include "cast/streaming/session_config.h" #include "platform/api/time.h" #include "util/yet_another_bit_vector.h" namespace openscreen { namespace cast { class Environment; // The Cast Streaming Sender, a peer corresponding to some Cast Streaming // Receiver at the other end of a network link. See class level comments for // Receiver for a high-level overview. // // The Sender is the peer responsible for enqueuing EncodedFrames for streaming, // guaranteeing their delivery to a Receiver, and handling feedback events from // a Receiver. Some feedback events are used for managing the Sender's internal // queue of in-flight frames, requesting network packet re-transmits, etc.; // while others are exposed via the Sender's public interface. For example, // sometimes the Receiver signals that it needs a a key frame to resolve a // picture loss condition, and the modules upstream of the Sender (e.g., where // encoding happens) should call NeedsKeyFrame() to check for, and handle that. // // There are usually one or two Senders in a streaming session, one for audio // and one for video. Both senders work with the same SenderPacketRouter // instance to schedule their transmission of packets, and provide the necessary // metrics for estimating bandwidth utilization and availability. // // It is the responsibility of upstream code modules to handle congestion // control. With respect to this Sender, that means the media encoding bit rate // should be throttled based on network bandwidth availability. This Sender does // not do any throttling, only flow-control. In other words, this Sender can // only manage its in-flight queue of frames, and if that queue grows too large, // it will eventually reject further enqueuing. // // General usage: A client should check the in-flight media duration frequently // to decide when to pause encoding, to avoid wasting system resources on // encoding frames that will likely be rejected by the Sender. The client should // also frequently call NeedsKeyFrame() and, when this returns true, direct its // encoder to produce a key frame soon. Finally, when using EnqueueFrame(), an // EncodedFrame struct should be prepared with its frame_id field set to // whatever GetNextFrameId() returns. Please see method comments for // more-detailed usage info. class Sender final : public SenderPacketRouter::Sender, public CompoundRtcpParser::Client { public: // Interface for receiving notifications about events of possible interest. // Handling each of these is optional, but some may be mandatory for certain // applications (see method comments below). class Observer { public: // Called when a frame was canceled. "Canceled" means that the Receiver has // either acknowledged successful receipt of the frame or has decided to // skip over it. Note: Frame cancellations may occur out-of-order. virtual void OnFrameCanceled(FrameId frame_id); // Called when a Receiver begins reporting picture loss, and there is no key // frame currently enqueued in the Sender. The application should enqueue a // key frame as soon as possible. Note: An application that pauses frame // sending (e.g., screen mirroring when the screen is not changing) should // use this notification to send an out-of-band "refresh frame," encoded as // a key frame. virtual void OnPictureLost(); protected: virtual ~Observer(); }; // Result codes for EnqueueFrame(). enum EnqueueFrameResult { // The frame has been queued for sending. OK, // The frame's payload was too large. This is typically triggered when // submitting a payload of several dozen megabytes or more. This result code // likely indicates some kind of upstream bug. PAYLOAD_TOO_LARGE, // The span of FrameIds is too large. Cast Streaming's protocol design // imposes a limit in the maximum difference between the highest-valued // in-flight FrameId and the least-valued one. REACHED_ID_SPAN_LIMIT, // Too-large a media duration is in-flight. Enqueuing another frame would // automatically cause late play-out at the Receiver. MAX_DURATION_IN_FLIGHT, }; // Constructs a Sender that attaches to the given |environment|-provided // resources and |packet_router|. The |config| contains the settings that were // agreed-upon by both sides from the OFFER/ANSWER exchange (i.e., the part of // the overall end-to-end connection process that occurs before Cast Streaming // is started). The |rtp_payload_type| does not affect the behavior of this // Sender. It is simply passed along to a Receiver in the RTP packet stream. Sender(Environment* environment, SenderPacketRouter* packet_router, SessionConfig config, RtpPayloadType rtp_payload_type); ~Sender() final; const SessionConfig& config() const { return config_; } Ssrc ssrc() const { return rtcp_session_.sender_ssrc(); } int rtp_timebase() const { return rtp_timebase_; } // Sets an observer for receiving notifications. Call with nullptr to stop // observing. void SetObserver(Observer* observer); // Returns the number of frames currently in-flight. This is only meant to be // informative. Clients should use GetInFlightMediaDuration() to make // throttling decisions. int GetInFlightFrameCount() const; // Returns the total media duration of the frames currently in-flight, // assuming the next not-yet-enqueued frame will have the given RTP timestamp. // For a better user experience, the result should be compared to // GetMaxInFlightMediaDuration(), and media encoding should be throttled down // before additional EnqueueFrame() calls would cause this to reach the // current maximum limit. Clock::duration GetInFlightMediaDuration( RtpTimeTicks next_frame_rtp_timestamp) const; // Return the maximum acceptable in-flight media duration, given the current // target playout delay setting and end-to-end network/system conditions. Clock::duration GetMaxInFlightMediaDuration() const; // Returns true if the Receiver requires a key frame. Note that this will // return true until a key frame is accepted by EnqueueFrame(). Thus, when // encoding is pipelined, care should be taken to instruct the encoder to // produce just ONE forced key frame. bool NeedsKeyFrame() const; // Returns the next FrameId, the one after the frame enqueued by the last call // to EnqueueFrame(). Note that the next call to EnqueueFrame() assumes this // frame ID be used. FrameId GetNextFrameId() const; // Enqueues the given |frame| for sending as soon as possible. Returns OK if // the frame is accepted, and some time later Observer::OnFrameCanceled() will // be called once it is no longer in-flight. // // All fields of the |frame| must be set to valid values: the |frame_id| must // be the same as GetNextFrameId(); both the |rtp_timestamp| and // |reference_time| fields must be monotonically increasing relative to the // prior frame; and the frame's |data| pointer must be set. [[nodiscard]] EnqueueFrameResult EnqueueFrame(const EncodedFrame& frame); // Causes all pending operations to discard data when they are processed // later. void CancelInFlightData(); private: // Tracking/Storage for frames that are ready-to-send, and until they are // fully received at the other end. struct PendingFrameSlot { // The frame to send, or nullopt if this slot is not in use. absl::optional frame; // Represents which packets need to be sent. Elements are indexed by // FramePacketId. A set bit means a packet needs to be sent (or re-sent). YetAnotherBitVector send_flags; // The time when each of the packets was last sent, or // |SenderPacketRouter::kNever| if the packet has not been sent yet. // Elements are indexed by FramePacketId. This is used to avoid // re-transmitting any given packet too frequently. std::vector packet_sent_times; PendingFrameSlot(); ~PendingFrameSlot(); bool is_active_for_frame(FrameId frame_id) const { return frame && frame->frame_id == frame_id; } }; // Return value from the ChooseXYZ() helper methods. struct ChosenPacket { PendingFrameSlot* slot = nullptr; FramePacketId packet_id{}; explicit operator bool() const { return !!slot; } }; // An extension of ChosenPacket that also includes the point-in-time when the // packet should be sent. struct ChosenPacketAndWhen : public ChosenPacket { Clock::time_point when = SenderPacketRouter::kNever; }; // SenderPacketRouter::Sender implementation. void OnReceivedRtcpPacket(Clock::time_point arrival_time, absl::Span packet) final; absl::Span GetRtcpPacketForImmediateSend( Clock::time_point send_time, absl::Span buffer) final; absl::Span GetRtpPacketForImmediateSend( Clock::time_point send_time, absl::Span buffer) final; Clock::time_point GetRtpResumeTime() final; // CompoundRtcpParser::Client implementation. void OnReceiverReferenceTimeAdvanced(Clock::time_point reference_time) final; void OnReceiverReport(const RtcpReportBlock& receiver_report) final; void OnReceiverIndicatesPictureLoss() final; void OnReceiverCheckpoint(FrameId frame_id, std::chrono::milliseconds playout_delay) final; void OnReceiverHasFrames(std::vector acks) final; void OnReceiverIsMissingPackets(std::vector nacks) final; // Helper to choose which packet to send, from those that have been flagged as // "need to send." Returns a "false" result if nothing needs to be sent. ChosenPacket ChooseNextRtpPacketNeedingSend(); // Helper that returns the packet that should be used to kick-start the // Receiver, and the time at which the packet should be sent. Returns a kNever // result if kick-starting is not needed. ChosenPacketAndWhen ChooseKickstartPacket(); // Cancels the given frame once it is known to have been fully received (i.e., // based on the ACK feedback from the Receiver in a RTCP packet). This clears // the corresponding entry in |pending_frames_| and notifies the Observer. void CancelPendingFrame(FrameId frame_id); // Inline helper to return the slot that would contain the tracking info for // the given |frame_id|. const PendingFrameSlot* get_slot_for(FrameId frame_id) const { return &pending_frames_[(frame_id - FrameId::first()) % pending_frames_.size()]; } PendingFrameSlot* get_slot_for(FrameId frame_id) { return &pending_frames_[(frame_id - FrameId::first()) % pending_frames_.size()]; } const SessionConfig config_; SenderPacketRouter* const packet_router_; RtcpSession rtcp_session_; CompoundRtcpParser rtcp_parser_; SenderReportBuilder sender_report_builder_; RtpPacketizer rtp_packetizer_; const int rtp_timebase_; FrameCrypto crypto_; // Ring buffer of PendingFrameSlots. The frame having FrameId x will always // be slotted at position x % pending_frames_.size(). Use get_slot_for() to // access the correct slot for a given FrameId. std::array pending_frames_{}; // A count of the number of frames in-flight (i.e., the number of active // entries in |pending_frames_|). int num_frames_in_flight_ = 0; // The ID of the last frame enqueued. FrameId last_enqueued_frame_id_ = FrameId::leader(); // Indicates that all of the packets for all frames up to and including this // FrameId have been successfully received (or otherwise do not need to be // re-transmitted). FrameId checkpoint_frame_id_ = FrameId::leader(); // The ID of the latest frame the Receiver seems to be aware of. FrameId latest_expected_frame_id_ = FrameId::leader(); // The target playout delay for the last-enqueued frame. This is auto-updated // when a frame is enqueued that changes the delay. std::chrono::milliseconds target_playout_delay_; FrameId playout_delay_change_at_frame_id_ = FrameId::first(); // The exact arrival time of the last RTCP packet. Clock::time_point rtcp_packet_arrival_time_ = SenderPacketRouter::kNever; // The near-term average round trip time. This is updated with each Sender // Report → Receiver Report round trip. This is initially zero, indicating the // round trip time has not been measured yet. Clock::duration round_trip_time_{0}; // Maintain current stats in a Sender Report that is ready for sending at any // time. This includes up-to-date lip-sync information, and packet and byte // count stats. RtcpSenderReport pending_sender_report_; // These are used to determine whether a key frame needs to be sent to the // Receiver. When the Receiver provides a picture loss notification, the // current checkpoint frame ID is stored in |picture_lost_at_frame_id_|. Then, // while |last_enqueued_key_frame_id_| is less than or equal to // |picture_lost_at_frame_id_|, the Sender knows it still needs to send a key // frame to resolve the picture loss condition. In all other cases, the // Receiver is either in a good state or is in the process of receiving the // key frame that will make that happen. FrameId picture_lost_at_frame_id_ = FrameId::leader(); FrameId last_enqueued_key_frame_id_ = FrameId::leader(); // The current observer (optional). Observer* observer_ = nullptr; }; } // namespace cast } // namespace openscreen #endif // CAST_STREAMING_SENDER_H_