Cast: Refactor RTCP handling

media/cast/net/cast_transport_sender.h

 // 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.
 
 // This is the main interface for the cast transport sender.  It accepts encoded
 // frames (both audio and video), encrypts their encoded data, packetizes them
 // and feeds them into a transport (e.g., UDP).
 
 // Construction of the Cast Sender and the Cast Transport Sender should be done
 // in the following order:
 // 1. Create CastTransportSender.
 // 2. Create CastSender (accepts CastTransportSender as an input).
-// 3. Call CastTransportSender::SetPacketReceiver to ensure that the packets
-//    received by the CastTransportSender will be sent to the CastSender.
-// Steps 3 can be done interchangeably.
 
 // Destruction: The CastTransportSender is assumed to be valid as long as the
 // CastSender is alive. Therefore the CastSender should be destructed before the
 // CastTransportSender.
-// This also works when the CastSender acts as a receiver for the RTCP packets
-// due to the weak pointers in the ReceivedPacket method in cast_sender_impl.cc.
 
 #ifndef MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_
 #define MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_
 
 #include "base/basictypes.h"
 #include "base/callback.h"
 #include "base/single_thread_task_runner.h"
 #include "base/threading/non_thread_safe.h"
 #include "base/time/tick_clock.h"
 #include "media/cast/logging/logging_defines.h"
 #include "media/cast/net/cast_transport_config.h"
 #include "media/cast/net/cast_transport_defines.h"
+#include "media/cast/net/rtcp/receiver_rtcp_event_subscriber.h"
+#include "media/cast/net/rtcp/rtcp_defines.h"
 #include "net/base/ip_endpoint.h"
 
 namespace net {
 class NetLog;
 }  // namespace net
 
 namespace media {
 namespace cast {
 
 // Following the initialization of either audio or video an initialization
 // status will be sent via this callback.
 typedef base::Callback<void(CastTransportStatus status)>
     CastTransportStatusCallback;
 
-typedef base::Callback<void(const std::vector<PacketEvent>&)>
+typedef base::Callback<void(const std::vector<PacketEvent>&,
+                            const std::vector<FrameEvent>&)>
     BulkRawEventsCallback;
 
 // The application should only trigger this class from the transport thread.
 class CastTransportSender : public base::NonThreadSafe {
  public:
   static scoped_ptr<CastTransportSender> Create(
       net::NetLog* net_log,
       base::TickClock* clock,
       const net::IPEndPoint& remote_end_point,
       const CastTransportStatusCallback& status_callback,
       const BulkRawEventsCallback& raw_events_callback,
       base::TimeDelta raw_events_callback_interval,
       const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner);
 
   virtual ~CastTransportSender() {}
 
   // Audio/Video initialization.
   // Encoded frames cannot be transmitted until the relevant initialize method
-  // is called. Usually called by CastSender.
-  virtual void InitializeAudio(const CastTransportRtpConfig& config) = 0;
-  virtual void InitializeVideo(const CastTransportRtpConfig& config) = 0;
-
-  // Sets the Cast packet receiver. Should be called after creation on the
-  // Cast sender. Packets won't be received until this function is called.
-  virtual void SetPacketReceiver(
-      const PacketReceiverCallback& packet_receiver) = 0;
+  // is called.
+  virtual void InitializeAudio(const CastTransportRtpConfig& config,
+                               const RtcpCastMessageCallback& cast_message_cb,
+                               const RtcpRttCallback& rtt_cb) = 0;
+  virtual void InitializeVideo(const CastTransportRtpConfig& config,
+                               const RtcpCastMessageCallback& cast_message_cb,
+                               const RtcpRttCallback& rtt_cb) = 0;
 
   // The following two functions handle the encoded media frames (audio and
   // video) to be processed.
   // Frames will be encrypted, packetized and transmitted to the network.
   virtual void InsertCodedAudioFrame(const EncodedFrame& audio_frame) = 0;
   virtual void InsertCodedVideoFrame(const EncodedFrame& video_frame) = 0;
 
-  // Builds an RTCP packet and sends it to the network.
-  // |ntp_seconds|, |ntp_fraction| and |rtp_timestamp| are used in the
-  // RTCP Sender Report.
-  virtual void SendRtcpFromRtpSender(uint32 packet_type_flags,
-                                     uint32 ntp_seconds,
-                                     uint32 ntp_fraction,
-                                     uint32 rtp_timestamp,
-                                     const RtcpDlrrReportBlock& dlrr,
-                                     uint32 sending_ssrc,
-                                     const std::string& c_name) = 0;
+  // Sends a RTCP sender report to the receiver.
+  // |ssrc| is the SSRC for this report.
+  // |current_time| is the current time reported by a tick clock.
+  // |current_time_as_rtp_timestamp| is the corresponding RTP timestamp.
+  virtual void SendSenderReport(
+      uint32 ssrc,
+      base::TimeTicks current_time,
+      uint32 current_time_as_rtp_timestamp) = 0;
 
   // Retransmission request.
   // |missing_packets| includes the list of frames and packets in each
   // frame to be re-transmitted.
   // If |cancel_rtx_if_not_in_list| is used as an optimization to cancel
   // pending re-transmission requests of packets not listed in
   // |missing_packets|. If the requested packet(s) were sent recently
   // (how long is specified by |dedupe_window|) then this re-transmit
   // will be ignored.
   virtual void ResendPackets(
       bool is_audio,
       const MissingFramesAndPacketsMap& missing_packets,
       bool cancel_rtx_if_not_in_list,
       base::TimeDelta dedupe_window) = 0;
+
+  // Returns a callback for receiving packets for testing purposes.
+  virtual PacketReceiverCallback PacketReceiverForTesting();
 };
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_