[Cast] Clean-up: Merge RtpReceiver+AudioReceiver+VideoReceiver-->FrameReceiver.

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.
-//
-// 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.
-class AudioReceiver : public RtpReceiver,
-                      public RtpPayloadFeedback,
-                      public base::NonThreadSafe,
-                      public base::SupportsWeakPtr<AudioReceiver> {
- public:
-  AudioReceiver(scoped_refptr<CastEnvironment> cast_environment,
-                const FrameReceiverConfig& audio_config,
-                transport::PacedPacketSender* const packet_sender);
-
-  virtual ~AudioReceiver();
-
-  // Request a decoded audio frame.  The audio signal data returned in the
-  // callback will have the sampling rate and number of channels as requested in
-  // the configuration that was passed to the ctor.
-  //
-  // The given |callback| is guaranteed to be run at some point in the future,
-  // even if to respond with NULL at shutdown time.
-  void GetRawAudioFrame(const AudioFrameDecodedCallback& callback);
-
-  // 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);
-
- 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);
-
-  // 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();
-
-  // Actually send the next cast message.
-  void SendNextCastMessage();
-
-  // Receives an AudioBus from |audio_decoder_|, logs the event, and passes the
-  // data on by running the given |callback|.  This method is static to ensure
-  // it can be called after an AudioReceiver instance is destroyed.
-  // DecodeEncodedAudioFrame() uses this as a callback for
-  // AudioDecoder::DecodeFrame().
-  static void EmitRawAudioFrame(
-      const scoped_refptr<CastEnvironment>& cast_environment,
-      const AudioFrameDecodedCallback& callback,
-      uint32 frame_id,
-      uint32 rtp_timestamp,
-      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_;
-
-  // 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_;
-
-  // Hack: This is used in logic that determines whether to skip frames.
-  const base::TimeDelta expected_frame_duration_;
-
-  // 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_;
-
-  // 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_