[Cast] In Audio/VideoSender, drop frames when too-long a duration is in-flight.

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/rtp_timestamp_helper.h"
 
 namespace media {
 namespace cast {
 
 class FrameSender {
  public:
   FrameSender(scoped_refptr<CastEnvironment> cast_environment,
               CastTransportSender* const transport_sender,
               base::TimeDelta rtcp_interval,
-              int frequency,
+              int rtp_timebase,
               uint32 ssrc,
               double max_frame_rate,
               base::TimeDelta playout_delay);
   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 {
@@ skipping 16 matching lines @@
 
   // 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 ssrc_;
 
-  // Records lip-sync (i.e., mapping of RTP <--> NTP timestamps), and
-  // extrapolates this mapping to any other point in time.
-  RtpTimestampHelper rtp_timestamp_helper_;
-
   // RTT information from RTCP.
   bool rtt_available_;
   base::TimeDelta rtt_;
   base::TimeDelta avg_rtt_;
   base::TimeDelta min_rtt_;
   base::TimeDelta max_rtt_;
 
  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();
 
+  // 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;
+
   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_;
 
   // 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 AudioSender assumes this
+  // 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_;
 
-  // This is a "good enough" mapping for finding the RTP timestamp associated
-  // with a video frame. The key is the lowest 8 bits of frame id (which is
-  // what is sent via RTCP). This map is used for logging purposes.
-  RtpTimestamp frame_id_to_rtp_timestamp_[256];
+ private:
+  // RTP timestamp increment representing one second.
+  const int rtp_timebase_;
 
- private:
+  // 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_