Cast: Move retransmission to the transport

media/cast/net/cast_transport_sender_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 send 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:
 //
 // CastTransportSender              RTP                      RTCP
 // ------------------------------------------------------------------
 //                      TransportEncryptionHandler (A/V)
 //                      RtpSender (A/V)                   Rtcp (A/V)
 //                                      PacedSender (Shared)
 //                                      UdpTransport (Shared)
 //
 // There are objects of TransportEncryptionHandler, RtpSender and Rtcp
 // for each audio and video stream.
 // PacedSender and UdpTransport are shared between all RTP and RTCP
 // streams.
 
 #ifndef MEDIA_CAST_NET_CAST_TRANSPORT_IMPL_H_
 #define MEDIA_CAST_NET_CAST_TRANSPORT_IMPL_H_
 
 #include "base/callback.h"
+#include "base/gtest_prod_util.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 "base/timer/timer.h"
 #include "media/cast/common/transport_encryption_handler.h"
 #include "media/cast/logging/logging_defines.h"
 #include "media/cast/logging/simple_event_subscriber.h"
 #include "media/cast/net/cast_transport_config.h"
@@ skipping 34 matching lines @@
                                const RtcpCastMessageCallback& cast_message_cb,
                                const RtcpRttCallback& rtt_cb) OVERRIDE;
   virtual void InsertCodedAudioFrame(const EncodedFrame& audio_frame) OVERRIDE;
   virtual void InsertCodedVideoFrame(const EncodedFrame& video_frame) OVERRIDE;
 
   virtual void SendSenderReport(
       uint32 ssrc,
       base::TimeTicks current_time,
       uint32 current_time_as_rtp_timestamp) OVERRIDE;
 
-  virtual void ResendPackets(bool is_audio,
-                             const MissingFramesAndPacketsMap& missing_packets,
-                             bool cancel_rtx_if_not_in_list,
-                             base::TimeDelta dedupe_window)
-      OVERRIDE;
+  virtual void CancelSendingFrames(uint32 ssrc,
+                                   const std::set<uint32>& frame_ids) OVERRIDE;
+
+  virtual void ResendFrameForKickstart(uint32 ssrc, uint32 frame_id) OVERRIDE;
 
   virtual PacketReceiverCallback PacketReceiverForTesting() OVERRIDE;
 
  private:
+  FRIEND_TEST_ALL_PREFIXES(CastTransportSenderImplTest, NacksCancelRetransmits);
+  FRIEND_TEST_ALL_PREFIXES(CastTransportSenderImplTest, CancelRetransmits);
+  FRIEND_TEST_ALL_PREFIXES(CastTransportSenderImplTest, Kickstart);
+
+  // 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.
+  // If packet was sent after |now - dedupe_window| then it will not be sent.
+  void ResendPackets(uint32 ssrc,
+                     const MissingFramesAndPacketsMap& missing_packets,
+                     bool cancel_rtx_if_not_in_list,
+                     base::TimeDelta dedupe_window);
+
   // If |raw_events_callback_| is non-null, calls it with events collected
   // by |event_subscriber_| since last call.
   void SendRawEvents();
 
   // Called when a packet is received.
   void OnReceivedPacket(scoped_ptr<Packet> packet);
 
   // Called when a log message is received.
   void OnReceivedLogMessage(EventMediaType media_type,
                             const RtcpReceiverLogMessage& log);
 
+  // Called when RTT information is updated.
+  void OnReceivedRtt(const RtcpRttCallback& rtt_cb,
+                     base::TimeDelta rtt,
+                     base::TimeDelta avg_rtt,
+                     base::TimeDelta min_rtt,
+                     base::TimeDelta max_rtt);
+
+  // Called when a RTCP Cast message is received.
+  void OnReceivedCastMessage(uint32 ssrc,
+                             const RtcpCastMessageCallback& cast_message_cb,
+                             const RtcpCastMessage& cast_message);
+
   base::TickClock* clock_;  // Not owned by this class.
   CastTransportStatusCallback status_callback_;
   scoped_refptr<base::SingleThreadTaskRunner> transport_task_runner_;
 
   LoggingImpl logging_;
 
   // Interface to a UDP socket.
   scoped_ptr<UdpTransport> transport_;
 
   // Packet sender that performs pacing.
@@ skipping 13 matching lines @@
   // the damage that could be caused by a compromised renderer process.
   TransportEncryptionHandler audio_encryptor_;
   TransportEncryptionHandler video_encryptor_;
 
   // This is non-null iff |raw_events_callback_| is non-null.
   scoped_ptr<SimpleEventSubscriber> event_subscriber_;
 
   BulkRawEventsCallback raw_events_callback_;
   base::TimeDelta raw_events_callback_interval_;
 
+  // The most up-to-date RTT information from either audio of video RTCP
+  // stream.
+  base::TimeDelta most_recent_rtt_;

[hubbe, 2014/08/11 19:11:50]

Given that audio packets are usually smaller than video packets, do we really
want to mix up their RTTs?

[Alpha Left Google, 2014/08/11 21:03:18]

RTT is gathered by RTCP packets so it doesn't matter how large the RTP payload
packets are. And they are muxed to the same socket anyway.

In the latest patchset I'm using RTT from each RTCP estimate.

+
   base::WeakPtrFactory<CastTransportSenderImpl> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(CastTransportSenderImpl);
 };
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_NET_CAST_TRANSPORT_IMPL_H_