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
  19 namespace media {
  20 namespace cast {
  21
  22 class FrameSender {
  23  public:
  24   FrameSender(scoped_refptr<CastEnvironment> cast_environment,
  25               CastTransportSender* const transport_sender,
  26               base::TimeDelta rtcp_interval,
  27               int rtp_timebase,
  28               uint32 ssrc,
  29               double max_frame_rate,
  30               base::TimeDelta playout_delay);
  31   virtual ~FrameSender();
  32
  33   // Calling this function is only valid if the receiver supports the
  34   // "extra_playout_delay", rtp extension.
  35   void SetTargetPlayoutDelay(base::TimeDelta new_target_playout_delay);
  36
  37   base::TimeDelta GetTargetPlayoutDelay() const {
  38     return target_playout_delay_;
  39   }
  40
  41  protected:
  42   // Schedule and execute periodic sending of RTCP report.
  43   void ScheduleNextRtcpReport();
  44   void SendRtcpReport(bool schedule_future_reports);
  45
  46   void OnReceivedRtt(base::TimeDelta rtt,
  47                      base::TimeDelta avg_rtt,
  48                      base::TimeDelta min_rtt,
  49                      base::TimeDelta max_rtt);
  50
  51   bool is_rtt_available() const { return rtt_available_; }
  52
  53   const scoped_refptr<CastEnvironment> cast_environment_;
  54
  55   // Sends encoded frames over the configured transport (e.g., UDP).  In
  56   // Chromium, this could be a proxy that first sends the frames from a renderer
  57   // process to the browser process over IPC, with the browser process being
  58   // responsible for "packetizing" the frames and pushing packets into the
  59   // network layer.
  60   CastTransportSender* const transport_sender_;
  61
  62   const uint32 ssrc_;
  63
  64   // RTT information from RTCP.
  65   bool rtt_available_;
  66   base::TimeDelta rtt_;
  67   base::TimeDelta avg_rtt_;
  68   base::TimeDelta min_rtt_;
  69   base::TimeDelta max_rtt_;
  70
  71  protected:
  72   // Schedule and execute periodic checks for re-sending packets.  If no
  73   // acknowledgements have been received for "too long," AudioSender will
  74   // speculatively re-send certain packets of an unacked frame to kick-start
  75   // re-transmission.  This is a last resort tactic to prevent the session from
  76   // getting stuck after a long outage.
  77   void ScheduleNextResendCheck();
  78   void ResendCheck();
  79   void ResendForKickstart();
  80
  81   // Record or retrieve a recent history of each frame's timestamps.
  82   // Warning: If a frame ID too far in the past is requested, the getters will
  83   // silently succeed but return incorrect values.  Be sure to respect
  84   // media::cast::kMaxUnackedFrames.
  85   void RecordLatestFrameTimestamps(uint32 frame_id,
  86                                    base::TimeTicks reference_time,
  87                                    RtpTimestamp rtp_timestamp);
  88   base::TimeTicks GetRecordedReferenceTime(uint32 frame_id) const;
  89   RtpTimestamp GetRecordedRtpTimestamp(uint32 frame_id) const;
  90
  91   const base::TimeDelta rtcp_interval_;
  92
  93   // The total amount of time between a frame's capture/recording on the sender
  94   // and its playback on the receiver (i.e., shown to a user).  This is fixed as
  95   // a value large enough to give the system sufficient time to encode,
  96   // transmit/retransmit, receive, decode, and render; given its run-time
  97   // environment (sender/receiver hardware performance, network conditions,
  98   // etc.).
  99   base::TimeDelta target_playout_delay_;
 100
 101   // If true, we transmit the target playout delay to the receiver.
 102   bool send_target_playout_delay_;
 103
 104   // Max encoded frames generated per second.
 105   double max_frame_rate_;
 106
 107   // Maximum number of outstanding frames before the encoding and sending of
 108   // new frames shall halt.
 109   int max_unacked_frames_;
 110
 111   // Counts how many RTCP reports are being "aggressively" sent (i.e., one per
 112   // frame) at the start of the session.  Once a threshold is reached, RTCP
 113   // reports are instead sent at the configured interval + random drift.
 114   int num_aggressive_rtcp_reports_sent_;
 115
 116   // This is "null" until the first frame is sent.  Thereafter, this tracks the
 117   // last time any frame was sent or re-sent.
 118   base::TimeTicks last_send_time_;
 119
 120   // The ID of the last frame sent.  Logic throughout FrameSender assumes this
 121   // can safely wrap-around.  This member is invalid until
 122   // |!last_send_time_.is_null()|.
 123   uint32 last_sent_frame_id_;
 124
 125   // The ID of the latest (not necessarily the last) frame that has been
 126   // acknowledged.  Logic throughout AudioSender assumes this can safely
 127   // wrap-around.  This member is invalid until |!last_send_time_.is_null()|.
 128   uint32 latest_acked_frame_id_;
 129
 130   // Counts the number of duplicate ACK that are being received.  When this
 131   // number reaches a threshold, the sender will take this as a sign that the
 132   // receiver hasn't yet received the first packet of the next frame.  In this
 133   // case, VideoSender will trigger a re-send of the next frame.
 134   int duplicate_ack_counter_;
 135
 136   // If this sender is ready for use, this is STATUS_AUDIO_INITIALIZED or
 137   // STATUS_VIDEO_INITIALIZED.
 138   CastInitializationStatus cast_initialization_status_;
 139
 140  private:
 141   // RTP timestamp increment representing one second.
 142   const int rtp_timebase_;
 143
 144   // Ring buffers to keep track of recent frame timestamps (both in terms of
 145   // local reference time and RTP media time).  These should only be accessed
 146   // through the Record/GetXXX() methods.
 147   base::TimeTicks frame_reference_times_[256];
 148   RtpTimestamp frame_rtp_timestamps_[256];
 149
 150   // NOTE: Weak pointers must be invalidated before all other member variables.
 151   base::WeakPtrFactory<FrameSender> weak_factory_;
 152
 153   DISALLOW_COPY_AND_ASSIGN(FrameSender);
 154 };
 155
 156 }  // namespace cast
 157 }  // namespace media
 158
 159 #endif  // MEDIA_CAST_SENDER_FRAME_SENDER_H_