Refactoring: CastTransport InitializeAudio/InitializeVideo.

media/cast/net/pacing/paced_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.
 
 #ifndef MEDIA_CAST_NET_PACING_PACED_SENDER_H_
 #define MEDIA_CAST_NET_PACING_PACED_SENDER_H_
 
 #include <stddef.h>
 #include <stdint.h>
 
@@ skipping 97 matching lines @@
       size_t target_burst_size,  // Should normally be kTargetBurstSize.
       size_t max_burst_size,     // Should normally be kMaxBurstSize.
       base::TickClock* clock,
       std::vector<PacketEvent>* recent_packet_events,
       PacketTransport* external_transport,
       const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner);
 
   ~PacedSender() final;
 
   // These must be called before non-RTCP packets are sent.
-  void RegisterAudioSsrc(uint32_t audio_ssrc);
-  void RegisterVideoSsrc(uint32_t video_ssrc);
+  void RegisterSsrc(uint32_t ssrc, bool is_audio);
 
   // Register SSRC that has a higher priority for sending. Multiple SSRCs can
   // be registered.
   // Note that it is not expected to register many SSRCs with this method.
   // Because IsHigherPriority() is determined in linear time.
   void RegisterPrioritySsrc(uint32_t ssrc);
 
   // Returns the total number of bytes sent to the socket when the specified
   // packet was just sent.
   // Returns 0 if the packet cannot be found or not yet sent.
+  // This function is currently only used by unittests.
   int64_t GetLastByteSentForPacket(const PacketKey& packet_key);
 
   // Returns the total number of bytes sent to the socket when the last payload
-  // identified by SSRC is just sent.
+  // identified by SSRC is just sent. Returns 0 for an unknown ssrc.
+  // This function is currently only used by unittests.
   int64_t GetLastByteSentForSsrc(uint32_t ssrc);
 
   // PacedPacketSender implementation.
   bool SendPackets(const SendPacketVector& packets) final;
   bool ResendPackets(const SendPacketVector& packets,
                      const DedupInfo& dedup_info) final;
   bool SendRtcpPacket(uint32_t ssrc, PacketRef packet) final;
   void CancelSendingPacket(const PacketKey& packet_key) final;
 
   void SetTargetBurstSize(int burst_size) {
@@ skipping 54 matching lines @@
 
   // Returns true if the packet should have a higher priority.
   bool IsHighPriority(const PacketKey& packet_key) const;
 
   // These are externally-owned objects injected via the constructor.
   base::TickClock* const clock_;
   std::vector<PacketEvent>* const recent_packet_events_;
   PacketTransport* const transport_;
 
   scoped_refptr<base::SingleThreadTaskRunner> transport_task_runner_;
-  uint32_t audio_ssrc_;
-  uint32_t video_ssrc_;
 
   // Set of SSRCs that have higher priority. This is a vector instead of a
   // set because there's only very few in it (most likely 1).
   std::vector<uint32_t> priority_ssrcs_;
   typedef std::map<PacketKey, std::pair<PacketType, PacketRef> > PacketList;
   PacketList packet_list_;
   PacketList priority_packet_list_;
 
-  struct PacketSendRecord {
-    PacketSendRecord();
-    base::TimeTicks time;  // Time when the packet was sent.
-    int64_t last_byte_sent;  // Number of bytes sent to network just after this
-                             // packet was sent.
-    int64_t last_byte_sent_for_audio;  // Number of bytes sent to network from
-                                       // audio stream just before this packet.
-    int cancel_count;  // Number of times the packet was canceled (debugging).
-  };
-  typedef std::map<PacketKey, PacketSendRecord> PacketSendHistory;
+  struct PacketSendRecord;
+  using PacketSendHistory = std::map<PacketKey, PacketSendRecord>;
   PacketSendHistory send_history_;
   PacketSendHistory send_history_buffer_;
-  // Records the last byte sent for payload with a specific SSRC.
-  std::map<uint32_t, int64_t> last_byte_sent_;
+
+  struct RtpSession;
+  using SessionMap = std::map<uint32_t, RtpSession>;
+  // Records all the cast sessions with the sender SSRC as the key. These
+  // sessions are in sync with those in CastTransportImpl.
+  SessionMap sessions_;
+
+  // Records the last byte sent for audio payload.
+  int64_t last_byte_sent_for_audio_;
 
   size_t target_burst_size_;
   size_t max_burst_size_;
 
   // Maximum burst size for the next three bursts.
   size_t current_max_burst_size_;
   size_t next_max_burst_size_;
   size_t next_next_max_burst_size_;
   // Number of packets already sent in the current burst.
   size_t current_burst_size_;
   // This is when the current burst ends.
   base::TimeTicks burst_end_;
 
   State state_;
 
   bool has_reached_upper_bound_once_;
 
-  // Tracks recently-logged RTP timestamps so that it can expand the truncated
-  // values found in packets.
-  RtpTimeTicks last_logged_audio_rtp_timestamp_;
-  RtpTimeTicks last_logged_video_rtp_timestamp_;
-
   // NOTE: Weak pointers must be invalidated before all other member variables.
   base::WeakPtrFactory<PacedSender> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(PacedSender);
 };
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_NET_PACING_PACED_SENDER_H_