media/cast/sender/frame_sender.h

   1 // Copyright 2014 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4 //
   5 // This is the base class for an object that send frames to a receiver.
   6 // TODO(hclam): Refactor such that there is no separate AudioSender vs.
   7 // VideoSender, and the functionality of both is rolled into this class.
   8
   9 #ifndef MEDIA_CAST_SENDER_FRAME_SENDER_H_
  10 #define MEDIA_CAST_SENDER_FRAME_SENDER_H_
  11
  12 #include "base/basictypes.h"
  13 #include "base/memory/ref_counted.h"
  14 #include "base/memory/weak_ptr.h"
  15 #include "base/time/time.h"
  16 #include "media/cast/cast_environment.h"
  17 #include "media/cast/net/rtcp/rtcp.h"
  18 #include "media/cast/sender/congestion_control.h"
  19
  20 namespace media {
  21 namespace cast {
  22
  23 class FrameSender {
  24  public:
  25   FrameSender(scoped_refptr<CastEnvironment> cast_environment,
  26               bool is_audio,
  27               CastTransportSender* const transport_sender,
  28               int rtp_timebase,
  29               uint32 ssrc,
  30               double max_frame_rate,
  31               base::TimeDelta min_playout_delay,
  32               base::TimeDelta max_playout_delay,
  33               CongestionControl* congestion_control);
  34   virtual ~FrameSender();
  35
  36   int rtp_timebase() const { return rtp_timebase_; }
  37
  38   // Calling this function is only valid if the receiver supports the
  39   // "extra_playout_delay", rtp extension.
  40   void SetTargetPlayoutDelay(base::TimeDelta new_target_playout_delay);
  41
  42   base::TimeDelta GetTargetPlayoutDelay() const {
  43     return target_playout_delay_;
  44   }
  45
  46   // Called by the encoder with the next EncodeFrame to send.
  47   void SendEncodedFrame(int requested_bitrate_before_encode,
  48                         scoped_ptr<EncodedFrame> encoded_frame);
  49
  50  protected:
  51   // Returns the number of frames in the encoder's backlog.
  52   virtual int GetNumberOfFramesInEncoder() const = 0;
  53
  54   // Returns the duration of the data in the encoder's backlog plus the duration
  55   // of sent, unacknowledged frames.
  56   virtual base::TimeDelta GetInFlightMediaDuration() const = 0;
  57
  58   // Called when we get an ACK for a frame.
  59   virtual void OnAck(uint32 frame_id) = 0;
  60
  61  protected:
  62   // Schedule and execute periodic sending of RTCP report.
  63   void ScheduleNextRtcpReport();
  64   void SendRtcpReport(bool schedule_future_reports);
  65
  66   void OnMeasuredRoundTripTime(base::TimeDelta rtt);
  67
  68   const scoped_refptr<CastEnvironment> cast_environment_;
  69
  70   // Sends encoded frames over the configured transport (e.g., UDP).  In
  71   // Chromium, this could be a proxy that first sends the frames from a renderer
  72   // process to the browser process over IPC, with the browser process being
  73   // responsible for "packetizing" the frames and pushing packets into the
  74   // network layer.
  75   CastTransportSender* const transport_sender_;
  76
  77   const uint32 ssrc_;
  78
  79  protected:
  80   // Schedule and execute periodic checks for re-sending packets.  If no
  81   // acknowledgements have been received for "too long," AudioSender will
  82   // speculatively re-send certain packets of an unacked frame to kick-start
  83   // re-transmission.  This is a last resort tactic to prevent the session from
  84   // getting stuck after a long outage.
  85   void ScheduleNextResendCheck();
  86   void ResendCheck();
  87   void ResendForKickstart();
  88
  89   // Protected for testability.
  90   void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback);
  91
  92   // Returns true if too many frames would be in-flight by encoding and sending
  93   // the next frame having the given |frame_duration|.
  94   bool ShouldDropNextFrame(base::TimeDelta frame_duration) const;
  95
  96   // Record or retrieve a recent history of each frame's timestamps.
  97   // Warning: If a frame ID too far in the past is requested, the getters will
  98   // silently succeed but return incorrect values.  Be sure to respect
  99   // media::cast::kMaxUnackedFrames.
 100   void RecordLatestFrameTimestamps(uint32 frame_id,
 101                                    base::TimeTicks reference_time,
 102                                    RtpTimestamp rtp_timestamp);
 103   base::TimeTicks GetRecordedReferenceTime(uint32 frame_id) const;
 104   RtpTimestamp GetRecordedRtpTimestamp(uint32 frame_id) const;
 105
 106   // Returns the number of frames that were sent but not yet acknowledged.
 107   int GetUnacknowledgedFrameCount() const;
 108
 109   // The total amount of time between a frame's capture/recording on the sender
 110   // and its playback on the receiver (i.e., shown to a user).  This is fixed as
 111   // a value large enough to give the system sufficient time to encode,
 112   // transmit/retransmit, receive, decode, and render; given its run-time
 113   // environment (sender/receiver hardware performance, network conditions,
 114   // etc.).
 115   base::TimeDelta target_playout_delay_;
 116   base::TimeDelta min_playout_delay_;
 117   base::TimeDelta max_playout_delay_;
 118
 119   // If true, we transmit the target playout delay to the receiver.
 120   bool send_target_playout_delay_;
 121
 122   // Max encoded frames generated per second.
 123   double max_frame_rate_;
 124
 125   // Counts how many RTCP reports are being "aggressively" sent (i.e., one per
 126   // frame) at the start of the session.  Once a threshold is reached, RTCP
 127   // reports are instead sent at the configured interval + random drift.
 128   int num_aggressive_rtcp_reports_sent_;
 129
 130   // This is "null" until the first frame is sent.  Thereafter, this tracks the
 131   // last time any frame was sent or re-sent.
 132   base::TimeTicks last_send_time_;
 133
 134   // The ID of the last frame sent.  Logic throughout FrameSender assumes this
 135   // can safely wrap-around.  This member is invalid until
 136   // |!last_send_time_.is_null()|.
 137   uint32 last_sent_frame_id_;
 138
 139   // The ID of the latest (not necessarily the last) frame that has been
 140   // acknowledged.  Logic throughout AudioSender assumes this can safely
 141   // wrap-around.  This member is invalid until |!last_send_time_.is_null()|.
 142   uint32 latest_acked_frame_id_;
 143
 144   // Counts the number of duplicate ACK that are being received.  When this
 145   // number reaches a threshold, the sender will take this as a sign that the
 146   // receiver hasn't yet received the first packet of the next frame.  In this
 147   // case, VideoSender will trigger a re-send of the next frame.
 148   int duplicate_ack_counter_;
 149
 150   // This object controls how we change the bitrate to make sure the
 151   // buffer doesn't overflow.
 152   scoped_ptr<CongestionControl> congestion_control_;
 153
 154   // The most recently measured round trip time.
 155   base::TimeDelta current_round_trip_time_;
 156
 157  private:
 158   // Returns the maximum media duration currently allowed in-flight.  This
 159   // fluctuates in response to the currently-measured network latency.
 160   base::TimeDelta GetAllowedInFlightMediaDuration() const;
 161
 162   // RTP timestamp increment representing one second.
 163   const int rtp_timebase_;
 164
 165   const bool is_audio_;
 166
 167   // Ring buffers to keep track of recent frame timestamps (both in terms of
 168   // local reference time and RTP media time).  These should only be accessed
 169   // through the Record/GetXXX() methods.
 170   base::TimeTicks frame_reference_times_[256];
 171   RtpTimestamp frame_rtp_timestamps_[256];
 172
 173   // NOTE: Weak pointers must be invalidated before all other member variables.
 174   base::WeakPtrFactory<FrameSender> weak_factory_;
 175
 176   DISALLOW_COPY_AND_ASSIGN(FrameSender);
 177 };
 178
 179 }  // namespace cast
 180 }  // namespace media
 181
 182 #endif  // MEDIA_CAST_SENDER_FRAME_SENDER_H_