Refactor: simplify interface of SenderRtcpSession and CastTransport.

media/cast/net/cast_transport_impl.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 class maintains a transport for audio and video in a Cast Streaming
 // session.
 // Audio, video frames and RTCP messages are submitted to this object
 // and then packetized and paced to the underlying UDP socket.
 //
 // The hierarchy of send transport in a Cast Streaming session:
@@ skipping 15 matching lines @@
 
 #include <stdint.h>
 
 #include <set>
 #include <vector>
 
 #include "base/callback.h"
 #include "base/gtest_prod_util.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/time/tick_clock.h"
 #include "base/time/time.h"
 #include "media/cast/common/transport_encryption_handler.h"
 #include "media/cast/logging/logging_defines.h"
 #include "media/cast/net/cast_transport.h"
 #include "media/cast/net/cast_transport_config.h"
 #include "media/cast/net/pacing/paced_sender.h"
 #include "media/cast/net/rtcp/rtcp_builder.h"
 #include "media/cast/net/rtcp/sender_rtcp_session.h"
 #include "media/cast/net/rtp/rtp_parser.h"
 #include "media/cast/net/rtp/rtp_sender.h"
 #include "net/base/network_interfaces.h"
 
 namespace media {
 namespace cast {
 
 class UdpTransport;
 
 class CastTransportImpl final : public CastTransport {
  public:
   CastTransportImpl(
       base::TickClock* clock,  // Owned by the caller.
       base::TimeDelta logging_flush_interval,
-      scoped_ptr<Client> client,
-      scoped_ptr<PacketTransport> transport,
+      std::unique_ptr<Client> client,
+      std::unique_ptr<PacketTransport> transport,
       const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner);
 
   ~CastTransportImpl() final;
 
   // CastTransport implementation for sending.
   void InitializeAudio(const CastTransportRtpConfig& config,
-                       const RtcpCastMessageCallback& cast_message_cb,
-                       const RtcpRttCallback& rtt_cb,
-                       const RtcpPliCallback& pli_cb) final;
+                       std::unique_ptr<RtpSenderRtcpClient> rtcp_client) final;
   void InitializeVideo(const CastTransportRtpConfig& config,
-                       const RtcpCastMessageCallback& cast_message_cb,
-                       const RtcpRttCallback& rtt_cb,
-                       const RtcpPliCallback& pli_cb) final;
+                       std::unique_ptr<RtpSenderRtcpClient> rtcp_client) final;
   void InsertFrame(uint32_t ssrc, const EncodedFrame& frame) final;
 
   void SendSenderReport(uint32_t ssrc,
                         base::TimeTicks current_time,
                         RtpTimeTicks current_time_as_rtp_timestamp) final;
 
   void CancelSendingFrames(uint32_t ssrc,
                            const std::vector<uint32_t>& frame_ids) final;
 
   void ResendFrameForKickstart(uint32_t ssrc, uint32_t frame_id) final;
@@ skipping 23 matching lines @@
                                         const RtcpTimeData& time_data) final;
   void AddCastFeedback(const RtcpCastMessage& cast_message,
                        base::TimeDelta target_delay) final;
   void AddPli(const RtcpPliMessage& pli_message) final;
   void AddRtcpEvents(
       const ReceiverRtcpEventSubscriber::RtcpEvents& rtcp_events) final;
   void AddRtpReceiverReport(
       const RtcpReportBlock& rtp_receiver_report_block) final;
   void SendRtcpFromRtpReceiver() final;
 
+  // Called when a log message is received.
+  void OnReceivedLogMessage(EventMediaType media_type,

[miu, 2016/04/15 23:14:39]

These don't need to be made public.  (See my comments for
Audio/Video/FrameSender below for details on how to do this.)

[xjz, 2016/04/20 01:09:03]

Done.

+                            const RtcpReceiverLogMessage& log);
+
+  // Called when a RTCP Cast message is received.
+  void OnReceivedCastMessage(uint32_t ssrc,
+                             const RtcpCastMessage& cast_message);
+
  private:
   FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, NacksCancelRetransmits);
   FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, CancelRetransmits);
   FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, Kickstart);
   FRIEND_TEST_ALL_PREFIXES(CastTransportImplTest, DedupRetransmissionWithAudio);
 
   // Resend packets for the stream identified by |ssrc|.
   // If |cancel_rtx_if_not_in_list| is true then transmission of packets for the
   // frames but not in the list will be dropped.
   // See PacedSender::ResendPackets() to see how |dedup_info| works.
   void ResendPackets(uint32_t ssrc,
                      const MissingFramesAndPacketsMap& missing_packets,
                      bool cancel_rtx_if_not_in_list,
                      const DedupInfo& dedup_info);
 
   // If |logging_flush_interval| is set, this is called at approximate periodic
   // intervals.
   void SendRawEvents();
 
   // Called when a packet is received.
-  bool OnReceivedPacket(scoped_ptr<Packet> packet);
-
-  // Called when a log message is received.
-  void OnReceivedLogMessage(EventMediaType media_type,
-                            const RtcpReceiverLogMessage& log);
-
-  // Called when a RTCP Cast message is received.
-  void OnReceivedCastMessage(uint32_t ssrc,
-                             const RtcpCastMessageCallback& cast_message_cb,
-                             const RtcpCastMessage& cast_message);
+  bool OnReceivedPacket(std::unique_ptr<Packet> packet);
 
   base::TickClock* const clock_;  // Not owned by this class.
   const base::TimeDelta logging_flush_interval_;
-  const scoped_ptr<Client> transport_client_;
-  const scoped_ptr<PacketTransport> transport_;
+  const std::unique_ptr<Client> transport_client_;
+  const std::unique_ptr<PacketTransport> transport_;
   const scoped_refptr<base::SingleThreadTaskRunner> transport_task_runner_;
 
   // FrameEvents and PacketEvents pending delivery via raw events callback.
   // Do not add elements to these when |logging_flush_interval| is
   // |base::TimeDelta()|.
   std::vector<FrameEvent> recent_frame_events_;
   std::vector<PacketEvent> recent_packet_events_;
 
   // Packet sender that performs pacing.
   PacedSender pacer_;
 
   // Packetizer for audio and video frames.
-  scoped_ptr<RtpSender> audio_sender_;
-  scoped_ptr<RtpSender> video_sender_;
+  std::unique_ptr<RtpSender> audio_sender_;
+  std::unique_ptr<RtpSender> video_sender_;
 
   // Maintains RTCP session for audio and video.
-  scoped_ptr<SenderRtcpSession> audio_rtcp_session_;
-  scoped_ptr<SenderRtcpSession> video_rtcp_session_;
+  std::unique_ptr<SenderRtcpSession> audio_rtcp_session_;
+  std::unique_ptr<SenderRtcpSession> video_rtcp_session_;
 
   // Encrypts data in EncodedFrames before they are sent.  Note that it's
   // important for the encryption to happen here, in code that would execute in
   // the main browser process, for security reasons.  This helps to mitigate
   // the damage that could be caused by a compromised renderer process.
   TransportEncryptionHandler audio_encryptor_;
   TransportEncryptionHandler video_encryptor_;
 
   // Right after a frame is sent we record the number of bytes sent to the
   // socket. We record the corresponding bytes sent for the most recent ACKed
   // audio packet.
   int64_t last_byte_acked_for_audio_;
 
   // Packets that don't match these sender ssrcs are ignored.
   std::set<uint32_t> valid_sender_ssrcs_;
 
   // While non-null, global WiFi behavior modifications are in effect. This is
   // used, for example, to turn off WiFi scanning that tends to interfere with
   // the reliability of UDP packet transmission.
-  scoped_ptr<net::ScopedWifiOptions> wifi_options_autoreset_;
+  std::unique_ptr<net::ScopedWifiOptions> wifi_options_autoreset_;
 
   // Do not initialize the |rtcp_builder_at_rtp_receiver_| if the RTP receiver
   // SSRC does not match these ssrcs. Only RTP receiver needs to register its
   // SSRC in this set.
   std::set<uint32_t> valid_rtp_receiver_ssrcs_;
 
-  scoped_ptr<RtcpBuilder> rtcp_builder_at_rtp_receiver_;
+  std::unique_ptr<RtcpBuilder> rtcp_builder_at_rtp_receiver_;
 
   base::WeakPtrFactory<CastTransportImpl> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(CastTransportImpl);
 };
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_NET_CAST_TRANSPORT_IMPL_H_