Cast: Make receiver use cast_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_SENDER_IMPL_H_
 #define MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_IMPL_H_
 
+#include <set>
+
 #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 "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"
 #include "media/cast/net/cast_transport_sender.h"
 #include "media/cast/net/pacing/paced_sender.h"
 #include "media/cast/net/rtcp/rtcp.h"
+#include "media/cast/net/rtp/rtp_parser.h"
 #include "media/cast/net/rtp/rtp_sender.h"
 
 namespace media {
 namespace cast {
 
 class UdpTransport;
 
 class CastTransportSenderImpl : public CastTransportSender {
  public:
   // |external_transport| is only used for testing.
   // |raw_events_callback|: Raw events will be returned on this callback
   // which will be invoked every |raw_events_callback_interval|.
   // This can be a null callback, i.e. if user is not interested in raw events.
   // |raw_events_callback_interval|: This can be |base::TimeDelta()| if
   // |raw_events_callback| is a null callback.
   // |options| contains optional settings for the transport, possible
   // keys are:
   //   "DSCP" (value ignored) - turns DSCP on
   //   "pacer_target_burst_size": int - specifies how many packets to send
   //                                    per 10 ms ideally.
   //   "pacer_max_burst_size": int - specifies how many pakcets to send
   //                                 per 10 ms, max
   //   "send_buffer_min_size": int - specifies the minimum socket send buffer
   //                                 size
   //   "disable_wifi_scan" (value ignored) - disable wifi scans while streaming
   //   "media_streaming_mode" (value ignored) - turn media streaming mode on
   // Note, these options may be ignored on some platforms.
+  // TODO(hubbe): Too many callbacks, replace with an interface.
   CastTransportSenderImpl(
       net::NetLog* net_log,
       base::TickClock* clock,
+      const net::IPEndPoint& local_end_point,
       const net::IPEndPoint& remote_end_point,
       scoped_ptr<base::DictionaryValue> options,
       const CastTransportStatusCallback& status_callback,
       const BulkRawEventsCallback& raw_events_callback,
       base::TimeDelta raw_events_callback_interval,
       const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner,
+      const PacketReceiverCallback& packet_callback,
       PacketSender* external_transport);
 
   ~CastTransportSenderImpl() override;
 
+  // CastTransportSender implementation.
   void InitializeAudio(const CastTransportRtpConfig& config,
                        const RtcpCastMessageCallback& cast_message_cb,
                        const RtcpRttCallback& rtt_cb) override;
   void InitializeVideo(const CastTransportRtpConfig& config,
                        const RtcpCastMessageCallback& cast_message_cb,
                        const RtcpRttCallback& rtt_cb) override;
   void InsertFrame(uint32 ssrc, const EncodedFrame& frame) override;
 
   void SendSenderReport(uint32 ssrc,
                         base::TimeTicks current_time,
                         uint32 current_time_as_rtp_timestamp) override;
 
   void CancelSendingFrames(uint32 ssrc,
                            const std::vector<uint32>& frame_ids) override;
 
   void ResendFrameForKickstart(uint32 ssrc, uint32 frame_id) override;
 
   PacketReceiverCallback PacketReceiverForTesting() override;
 
+  // CastTransportReceiver implementation.
+  void AddValidSsrc(uint32 ssrc) override;
+
+  void SendRtcpFromRtpReceiver(
+      uint32 ssrc,
+      uint32 sender_ssrc,
+      const RtcpTimeData& time_data,
+      const RtcpCastMessage* cast_message,
+      base::TimeDelta target_delay,
+      const ReceiverRtcpEventSubscriber::RtcpEvents* rtcp_events,
+      const RtpReceiverStatistics* rtp_receiver_statistics) override;
+
  private:
   FRIEND_TEST_ALL_PREFIXES(CastTransportSenderImplTest, NacksCancelRetransmits);
   FRIEND_TEST_ALL_PREFIXES(CastTransportSenderImplTest, CancelRetransmits);
   FRIEND_TEST_ALL_PREFIXES(CastTransportSenderImplTest, Kickstart);
   FRIEND_TEST_ALL_PREFIXES(CastTransportSenderImplTest,
                            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 ssrc,
                      const MissingFramesAndPacketsMap& missing_packets,
                      bool cancel_rtx_if_not_in_list,
                      const DedupInfo& dedup_info);
 
   // 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);
+  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 ssrc,
                              const RtcpCastMessageCallback& cast_message_cb,
                              const RtcpCastMessage& cast_message);
 
@@ skipping 28 matching lines @@
   scoped_ptr<SimpleEventSubscriber> event_subscriber_;
 
   BulkRawEventsCallback raw_events_callback_;
   base::TimeDelta raw_events_callback_interval_;
 
   // 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 last_byte_acked_for_audio_;
 
+  // Packets that don't match these ssrcs are ignored.
+  std::set<uint32> valid_ssrcs_;
+
+  // Called with incoming packets. (Unless they match the
+  // channels created by Initialize{Audio,Video}.
+  PacketReceiverCallback packet_callback_;
+
   scoped_ptr<net::ScopedWifiOptions> wifi_options_autoreset_;
 
   base::WeakPtrFactory<CastTransportSenderImpl> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(CastTransportSenderImpl);
 };
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_IMPL_H_