[Cast] Repair receiver playout time calculations and frame skip logic.

media/cast/audio_receiver/audio_receiver.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_RECEIVER_AUDIO_RECEIVER_H_
 #define MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_
 
 #include "base/basictypes.h"
 #include "base/callback.h"
 #include "base/macros.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/cast/base/clock_drift_smoother.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/cast_receiver.h"
 #include "media/cast/framer/framer.h"
 #include "media/cast/rtcp/receiver_rtcp_event_subscriber.h"
 #include "media/cast/rtcp/rtcp.h"
 #include "media/cast/rtp_receiver/rtp_receiver.h"
 #include "media/cast/rtp_receiver/rtp_receiver_defines.h"
 #include "media/cast/transport/utility/transport_encryption_handler.h"
 
 namespace media {
 namespace cast {
 
 class AudioDecoder;
 
 // AudioReceiver receives packets out-of-order while clients make requests for
 // complete frames in-order.  (A frame consists of one or more packets.)
 //
 // AudioReceiver also includes logic for computing the playout time for each
 // frame, accounting for a constant targeted playout delay.  The purpose of the
 // playout delay is to provide a fixed window of time between the capture event
 // on the sender and the playout on the receiver.  This is important because
 // each step of the pipeline (i.e., encode frame, then transmit/retransmit from
 // the sender, then receive and re-order packets on the receiver, then decode
 // frame) can vary in duration and is typically very hard to predict.
-// Heuristics will determine when the targeted playout delay is insufficient in
-// the current environment; and the receiver can then increase the playout
-// delay, notifying the sender, to account for the extra variance.
-// TODO(miu): Make the last sentence true.  http://crbug.com/360111
 //
 // Two types of frames can be requested: 1) A frame of decoded audio data; or 2)
 // a frame of still-encoded audio data, to be passed into an external audio
 // decoder.  Each request for a frame includes a callback which AudioReceiver
 // guarantees will be called at some point in the future unless the
 // AudioReceiver is destroyed.  Clients should generally limit the number of
 // outstanding requests (perhaps to just one or two).
 //
 // This class is not thread safe.  Should only be called from the Main cast
 // thread.
@@ skipping 18 matching lines @@
 
   // Request an encoded audio frame.
   //
   // The given |callback| is guaranteed to be run at some point in the future,
   // even if to respond with NULL at shutdown time.
   void GetEncodedAudioFrame(const FrameEncodedCallback& callback);
 
   // Deliver another packet, possibly a duplicate, and possibly out-of-order.
   void IncomingPacket(scoped_ptr<Packet> packet);
 
-  // Update target audio delay used to compute the playout time. Rtcp
-  // will also be updated (will be included in all outgoing reports).
-  void SetTargetDelay(base::TimeDelta target_delay);
-
  protected:
   friend class AudioReceiverTest;  // Invokes OnReceivedPayloadData().
 
   virtual void OnReceivedPayloadData(const uint8* payload_data,
                                      size_t payload_size,
                                      const RtpCastHeader& rtp_header) OVERRIDE;
 
   // RtpPayloadFeedback implementation.
   virtual void CastFeedback(const RtcpCastMessage& cast_message) OVERRIDE;
 
  private:
   // Processes ready-to-consume packets from |framer_|, decrypting each packet's
   // payload data, and then running the enqueued callbacks in order (one for
   // each packet).  This method may post a delayed task to re-invoke itself in
   // the future to wait for missing/incomplete frames.
   void EmitAvailableEncodedFrames();
 
   // Clears the |is_waiting_for_consecutive_frame_| flag and invokes
   // EmitAvailableEncodedFrames().
   void EmitAvailableEncodedFramesAfterWaiting();
 
   // Feeds an EncodedFrame into |audio_decoder_|.  GetRawAudioFrame() uses this
   // as a callback for GetEncodedAudioFrame().
   void DecodeEncodedAudioFrame(
       const AudioFrameDecodedCallback& callback,
       scoped_ptr<transport::EncodedFrame> encoded_frame);
 
-  // Return the playout time based on the current time and rtp timestamp.
-  base::TimeTicks GetPlayoutTime(base::TimeTicks now, uint32 rtp_timestamp);
-
-  void InitializeTimers();
+  // Computes the playout time for a frame with the given |rtp_timestamp|.
+  // Because lip-sync info is refreshed regularly, calling this method with the
+  // same argument may return different results.
+  base::TimeTicks GetPlayoutTime(uint32 rtp_timestamp) const;
 
   // Schedule the next RTCP report.
   void ScheduleNextRtcpReport();
 
   // Actually send the next RTCP report.
   void SendNextRtcpReport();
 
   // Schedule timing for the next cast message.
   void ScheduleNextCastMessage();
 
@@ skipping 13 matching lines @@
       const base::TimeTicks& playout_time,
       scoped_ptr<AudioBus> audio_bus,
       bool is_continuous);
 
   const scoped_refptr<CastEnvironment> cast_environment_;
 
   // Subscribes to raw events.
   // Processes raw audio events to be sent over to the cast sender via RTCP.
   ReceiverRtcpEventSubscriber event_subscriber_;
 
+  // Configured audio codec.
   const transport::AudioCodec codec_;
+
+  // RTP timebase: The number of RTP units advanced per one second.  For audio,
+  // this is the sampling rate.
   const int frequency_;
-  base::TimeDelta target_delay_delta_;
+
+  // 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_;
+
+  // Set to false initially, then set to true after scheduling the periodic
+  // sending of reports back to the sender.  Reports are first scheduled just
+  // after receiving a first packet (since the first packet identifies the
+  // sender for the remainder of the session).
+  bool reports_are_scheduled_;
+
+  // Assembles packets into frames, providing this receiver with complete,
+  // decodable EncodedFrames.
   Framer framer_;
+
+  // Decodes frames into raw audio for playback.
   scoped_ptr<AudioDecoder> audio_decoder_;
+
+  // Manages sending/receiving of RTCP packets, including sender/receiver
+  // reports.
   Rtcp rtcp_;
-  base::TimeDelta time_offset_;
-  base::TimeTicks time_first_incoming_packet_;
-  uint32 first_incoming_rtp_timestamp_;
+
+  // Decrypts encrypted frames.
   transport::TransportEncryptionHandler decryptor_;
 
   // Outstanding callbacks to run to deliver on client requests for frames.
   std::list<FrameEncodedCallback> frame_request_queue_;
 
   // True while there's an outstanding task to re-invoke
   // EmitAvailableEncodedFrames().
   bool is_waiting_for_consecutive_frame_;
 
   // This mapping allows us to log AUDIO_ACK_SENT as a frame event. In addition
   // it allows the event to be transmitted via RTCP.
   RtpTimestamp frame_id_to_rtp_timestamp_[256];
 
+  // Lip-sync values used to compute the playout time of each frame from its RTP
+  // timestamp.  These are updated each time the first packet of a frame is
+  // received.
+  RtpTimestamp lip_sync_rtp_timestamp_;
+  base::TimeTicks lip_sync_reference_time_;
+  ClockDriftSmoother lip_sync_drift_;
+
   // NOTE: Weak pointers must be invalidated before all other member variables.
   base::WeakPtrFactory<AudioReceiver> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioReceiver);
 };
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_AUDIO_RECEIVER_AUDIO_RECEIVER_H_