[Cast] Halt AudioSender transmission when too many frames are in-flight.

media/cast/audio_sender/audio_sender.h

 // Copyright 2013 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.
 
 #ifndef MEDIA_CAST_AUDIO_SENDER_H_
 #define MEDIA_CAST_AUDIO_SENDER_H_
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/memory/weak_ptr.h"
 #include "base/threading/non_thread_safe.h"
 #include "base/time/tick_clock.h"
 #include "base/time/time.h"
 #include "media/base/audio_bus.h"
 #include "media/cast/cast_config.h"
+#include "media/cast/cast_environment.h"
+#include "media/cast/logging/logging_defines.h"
 #include "media/cast/rtcp/rtcp.h"
 #include "media/cast/rtp_timestamp_helper.h"
 
 namespace media {
 namespace cast {
 
 class AudioEncoder;
 
-// This class is not thread safe.
-// It's only called from the main cast thread.
+// Not thread safe. Only called from the main cast thread.
+// This class owns all objects related to sending audio, objects that create RTP
+// packets, congestion control, audio encoder, parsing and sending of
+// RTCP packets.
+// Additionally it posts a bunch of delayed tasks to the main thread for various
+// timeouts.
 class AudioSender : public RtcpSenderFeedback,
                     public base::NonThreadSafe,
                     public base::SupportsWeakPtr<AudioSender> {
  public:
   AudioSender(scoped_refptr<CastEnvironment> cast_environment,
               const AudioSenderConfig& audio_config,
               transport::CastTransportSender* const transport_sender);
 
   virtual ~AudioSender();
 
   CastInitializationStatus InitializationResult() const {
     return cast_initialization_status_;
   }
 
+  // Note: It is not guaranteed that |audio_frame| will actually be encoded and
+  // sent, if AudioSender detects too many frames in flight.  Therefore, clients
+  // should be careful about the rate at which this method is called.
+  //
   // Note: It is invalid to call this method if InitializationResult() returns
   // anything but STATUS_AUDIO_INITIALIZED.
   void InsertAudio(scoped_ptr<AudioBus> audio_bus,
                    const base::TimeTicks& recorded_time);
 
   // Only called from the main cast thread.
   void IncomingRtcpPacket(scoped_ptr<Packet> packet);
 
+ protected:
+  // Protected for testability.
+  virtual void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback)
+      OVERRIDE;
+
  private:
-  void ResendPackets(
-      const MissingFramesAndPacketsMap& missing_frames_and_packets);
-
+  // Schedule and execute periodic sending of RTCP report.
   void ScheduleNextRtcpReport();
   void SendRtcpReport(bool schedule_future_reports);
 
+  // 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();
+
+  // Returns true if there are too many frames in flight, as defined by the
+  // configured target playout delay plus simple logic.  When this is true,
+  // InsertAudio() will silenty drop frames instead of sending them to the audio
+  // encoder.
+  bool AreTooManyFramesInFlight() const;
+
   // Called by the |audio_encoder_| with the next EncodedFrame to send.
   void SendEncodedAudioFrame(scoped_ptr<transport::EncodedFrame> audio_frame);
 
-  virtual void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback)
-      OVERRIDE;
+  const scoped_refptr<CastEnvironment> cast_environment_;
 
-  scoped_refptr<CastEnvironment> cast_environment_;
+  // 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.).
+  const base::TimeDelta target_playout_delay_;
+
+  // 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.
   transport::CastTransportSender* const transport_sender_;
+
+  // Maximum number of outstanding frames before the encoding and sending of
+  // new frames shall halt.
+  const int max_unacked_frames_;
+
+  // Encodes AudioBuses into EncodedFrames.
   scoped_ptr<AudioEncoder> audio_encoder_;
+  const int configured_encoder_bitrate_;
+
+  // Manages sending/receiving of RTCP packets, including sender/receiver
+  // reports.
+  Rtcp rtcp_;
+
+  // Records lip-sync (i.e., mapping of RTP <--> NTP timestamps), and
+  // extrapolates this mapping to any other point in time.
   RtpTimestampHelper rtp_timestamp_helper_;
-  Rtcp rtcp_;
+
+  // 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
+  // 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, AudioSender 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.
   CastInitializationStatus cast_initialization_status_;
 
-  // Used to map the lower 8 bits of the frame id to a RTP timestamp. This is
-  // good enough as we only use it for logging.
+  // 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];
 
   // NOTE: Weak pointers must be invalidated before all other member variables.
   base::WeakPtrFactory<AudioSender> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioSender);
 };
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_AUDIO_SENDER_H_