// Copyright 2014 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. // // This is the base class for an object that send frames to a receiver. // TODO(hclam): Refactor such that there is no separate AudioSender vs. // VideoSender, and the functionality of both is rolled into this class. #ifndef MEDIA_CAST_SENDER_FRAME_SENDER_H_ #define MEDIA_CAST_SENDER_FRAME_SENDER_H_ #include <stdint.h> #include "base/macros.h" #include "base/memory/ref_counted.h" #include "base/memory/weak_ptr.h" #include "base/time/time.h" #include "media/cast/cast_environment.h" #include "media/cast/net/cast_transport_sender.h" #include "media/cast/net/rtcp/rtcp_defines.h" #include "media/cast/sender/congestion_control.h" namespace media { namespace cast { struct SenderEncodedFrame; class FrameSender { public: FrameSender(scoped_refptr<CastEnvironment> cast_environment, bool is_audio, CastTransportSender* const transport_sender, int rtp_timebase, uint32_t ssrc, double max_frame_rate, base::TimeDelta min_playout_delay, base::TimeDelta max_playout_delay, base::TimeDelta animated_playout_delay, CongestionControl* congestion_control); virtual ~FrameSender(); int rtp_timebase() const { return rtp_timebase_; } // Calling this function is only valid if the receiver supports the // "extra_playout_delay", rtp extension. void SetTargetPlayoutDelay(base::TimeDelta new_target_playout_delay); base::TimeDelta GetTargetPlayoutDelay() const { return target_playout_delay_; } // Called by the encoder with the next EncodeFrame to send. void SendEncodedFrame(int requested_bitrate_before_encode, scoped_ptr<SenderEncodedFrame> encoded_frame); protected: // Returns the number of frames in the encoder's backlog. virtual int GetNumberOfFramesInEncoder() const = 0; // Returns the duration of the data in the encoder's backlog plus the duration // of sent, unacknowledged frames. virtual base::TimeDelta GetInFlightMediaDuration() const = 0; protected: // Schedule and execute periodic sending of RTCP report. void ScheduleNextRtcpReport(); void SendRtcpReport(bool schedule_future_reports); void OnMeasuredRoundTripTime(base::TimeDelta rtt); const scoped_refptr<CastEnvironment> cast_environment_; // Sends encoded frames over the configured transport (e.g., UDP). In // Chromium, this could be a proxy that first sends the frames from a renderer // process to the browser process over IPC, with the browser process being // responsible for "packetizing" the frames and pushing packets into the // network layer. CastTransportSender* const transport_sender_; const uint32_t ssrc_; protected: // Schedule and execute periodic checks for re-sending packets. If no // acknowledgements have been received for "too long," AudioSender will // speculatively re-send certain packets of an unacked frame to kick-start // re-transmission. This is a last resort tactic to prevent the session from // getting stuck after a long outage. void ScheduleNextResendCheck(); void ResendCheck(); void ResendForKickstart(); // Protected for testability. void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback); // Returns true if too many frames would be in-flight by encoding and sending // the next frame having the given |frame_duration|. bool ShouldDropNextFrame(base::TimeDelta frame_duration) const; // Record or retrieve a recent history of each frame's timestamps. // Warning: If a frame ID too far in the past is requested, the getters will // silently succeed but return incorrect values. Be sure to respect // media::cast::kMaxUnackedFrames. void RecordLatestFrameTimestamps(uint32_t frame_id, base::TimeTicks reference_time, RtpTimeTicks rtp_timestamp); base::TimeTicks GetRecordedReferenceTime(uint32_t frame_id) const; RtpTimeTicks GetRecordedRtpTimestamp(uint32_t frame_id) const; // Returns the number of frames that were sent but not yet acknowledged. int GetUnacknowledgedFrameCount() const; // Playout delay represents total amount of time between a frame's // capture/recording on the sender and its playback on the receiver // (i.e., shown to a user). This should be a value large enough to // give the system sufficient time to encode, transmit/retransmit, // receive, decode, and render; given its run-time environment // (sender/receiver hardware performance, network conditions,etc.). // The |target_playout delay_| is the current delay that is adaptively // adjusted based on feedback from video capture engine and the congestion // control. In case of interactive content, the target is adjusted to start // at |min_playout_delay_| and in case of animated content, it starts out at // |animated_playout_delay_| and then adaptively adjust based on feedback // from congestion control. base::TimeDelta target_playout_delay_; const base::TimeDelta min_playout_delay_; const base::TimeDelta max_playout_delay_; // Starting playout delay for animated content. const base::TimeDelta animated_playout_delay_; // If true, we transmit the target playout delay to the receiver. bool send_target_playout_delay_; // Max encoded frames generated per second. double max_frame_rate_; // Counts how many RTCP reports are being "aggressively" sent (i.e., one per // frame) at the start of the session. Once a threshold is reached, RTCP // reports are instead sent at the configured interval + random drift. int num_aggressive_rtcp_reports_sent_; // This is "null" until the first frame is sent. Thereafter, this tracks the // last time any frame was sent or re-sent. base::TimeTicks last_send_time_; // The ID of the last frame sent. Logic throughout FrameSender assumes this // can safely wrap-around. This member is invalid until // |!last_send_time_.is_null()|. uint32_t last_sent_frame_id_; // The ID of the latest (not necessarily the last) frame that has been // acknowledged. Logic throughout AudioSender assumes this can safely // wrap-around. This member is invalid until |!last_send_time_.is_null()|. uint32_t latest_acked_frame_id_; // Counts the number of duplicate ACK that are being received. When this // number reaches a threshold, the sender will take this as a sign that the // receiver hasn't yet received the first packet of the next frame. In this // case, VideoSender will trigger a re-send of the next frame. int duplicate_ack_counter_; // This object controls how we change the bitrate to make sure the // buffer doesn't overflow. scoped_ptr<CongestionControl> congestion_control_; // The most recently measured round trip time. base::TimeDelta current_round_trip_time_; private: // Returns the maximum media duration currently allowed in-flight. This // fluctuates in response to the currently-measured network latency. base::TimeDelta GetAllowedInFlightMediaDuration() const; // RTP timestamp increment representing one second. const int rtp_timebase_; const bool is_audio_; // Ring buffers to keep track of recent frame timestamps (both in terms of // local reference time and RTP media time). These should only be accessed // through the Record/GetXXX() methods. base::TimeTicks frame_reference_times_[256]; RtpTimeTicks frame_rtp_timestamps_[256]; // NOTE: Weak pointers must be invalidated before all other member variables. base::WeakPtrFactory<FrameSender> weak_factory_; DISALLOW_COPY_AND_ASSIGN(FrameSender); }; } // namespace cast } // namespace media #endif // MEDIA_CAST_SENDER_FRAME_SENDER_H_