Cast: Reshuffle files under media/cast

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;
-
-// 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:
-  // 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);
-
-  const 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_;
-
-  // 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_;
-
-  // 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_