Cast: Reshuffle files under media/cast

This is a large refactoring to move files around where it makes sense.

media/cast/transport/rtp_sender => media/cast/net/rtp
media/cast/transport/rtp_sender/packet_storage => media/cast/net/rtp
media/cast/transport/rtp_sender/packetizer => media/cast/net/rtp
media/cast/transport/transport => media/cast/net
media/cast/base => media/cast/common
media/cast/transport/utility => media/cast/common
media/cast/video_sender => media/cast/sender
media/cast/video_sender/codecs/vp8 => media/cast/sender
media/cast/audio_sender => media/cast/sender
media/cast/congestion_control => media/cast/sender
media/cast/transport => media/cast/net
media/cast/framer => media/cast/net/rtp
media/cast/rtp_receiver => media/cast/net/rtp
media/cast/rtcp => media/cast/net/rtcp
Remove "transport" namespace

BUG=393042

Review URL: https://codereview.chromium.org/388663003

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@283120 0039d316-1c4b-4281-b951-d872f2087c98

1 files changed, 162 insertions, 0 deletions

diff --git a/media/cast/sender/audio_sender.h b/media/cast/sender/audio_sender.h
new file mode 100644
index 0000000..efaa2b3
--- /dev/null
+++ b/media/cast/sender/audio_sender.h
@@ -0,0 +1,162 @@
+// 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.
+
+#ifndef MEDIA_CAST_SENDER_AUDIO_SENDER_H_
+#define MEDIA_CAST_SENDER_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/net/rtcp/rtcp.h"
+#include "media/cast/sender/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,
+              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<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.
+  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_SENDER_AUDIO_SENDER_H_