Cast: Merge common functionality from audio/video sender into frame_sender.

media/cast/sender/frame_sender.h

 // 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 "base/basictypes.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/rtcp/rtcp.h"
+#include "media/cast/sender/congestion_control.h"
 
 namespace media {
 namespace cast {
 
 class FrameSender {
  public:
   FrameSender(scoped_refptr<CastEnvironment> cast_environment,
+              bool is_audio,
               CastTransportSender* const transport_sender,
               base::TimeDelta rtcp_interval,
               int rtp_timebase,
               uint32 ssrc,
               double max_frame_rate,
-              base::TimeDelta playout_delay);
+              base::TimeDelta playout_delay,
+              CongestionControl* congestion_control);
   virtual ~FrameSender();
 
   // 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<EncodedFrame> encoded_frame);
+
  protected:
   // Schedule and execute periodic sending of RTCP report.
   void ScheduleNextRtcpReport();
   void SendRtcpReport(bool schedule_future_reports);
 
   void OnReceivedRtt(base::TimeDelta rtt,
                      base::TimeDelta avg_rtt,
                      base::TimeDelta min_rtt,
                      base::TimeDelta max_rtt);
 
@@ skipping 20 matching lines @@
  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 there are too many frames in flight, or if the media
+  // duration of the frames in flight would be too high by sending the next
+  // frame.  The latter metric is determined from the given |capture_time|
+  // for the next frame to be encoded and sent.
+  bool ShouldDropNextFrame(base::TimeTicks capture_time) 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 frame_id,
                                    base::TimeTicks reference_time,
                                    RtpTimestamp rtp_timestamp);
   base::TimeTicks GetRecordedReferenceTime(uint32 frame_id) const;
   RtpTimestamp GetRecordedRtpTimestamp(uint32 frame_id) const;
 
+  // Called when we get an ACK for a frame.
+  virtual void OnAck(uint32 frame_id) = 0;
+
   const base::TimeDelta rtcp_interval_;
 
   // The 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 is fixed as
   // 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.).
   base::TimeDelta target_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_;
 
   // Maximum number of outstanding frames before the encoding and sending of
   // new frames shall halt.
   int max_unacked_frames_;
 
+  // The number of frames currently being processed in |video_encoder_|.
+  int frames_in_encoder_;
+
   // 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 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 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_;
 
   // If this sender is ready for use, this is STATUS_AUDIO_INITIALIZED or
   // STATUS_VIDEO_INITIALIZED.
   CastInitializationStatus cast_initialization_status_;
 
- private:
   // RTP timestamp increment representing one second.
   const int rtp_timebase_;
 
+  // This object controls how we change the bitrate to make sure the
+  // buffer doesn't overflow.
+  scoped_ptr<CongestionControl> congestion_control_;
+
+ private:
+  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];
   RtpTimestamp 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_