media/cast/net/cast_transport_sender_impl.h

   1 // Copyright 2014 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 // This class maintains a send transport for audio and video in a Cast
   6 // Streaming session.
   7 // Audio, video frames and RTCP messages are submitted to this object
   8 // and then packetized and paced to the underlying UDP socket.
   9 //
  10 // The hierarchy of send transport in a Cast Streaming session:
  11 //
  12 // CastTransportSender              RTP                      RTCP
  13 // ------------------------------------------------------------------
  14 //                      TransportEncryptionHandler (A/V)
  15 //                      RtpSender (A/V)                   Rtcp (A/V)
  16 //                                      PacedSender (Shared)
  17 //                                      UdpTransport (Shared)
  18 //
  19 // There are objects of TransportEncryptionHandler, RtpSender and Rtcp
  20 // for each audio and video stream.
  21 // PacedSender and UdpTransport are shared between all RTP and RTCP
  22 // streams.
  23
  24 #ifndef MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_IMPL_H_
  25 #define MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_IMPL_H_
  26
  27 #include <set>
  28
  29 #include "base/callback.h"
  30 #include "base/gtest_prod_util.h"
  31 #include "base/memory/ref_counted.h"
  32 #include "base/memory/scoped_ptr.h"
  33 #include "base/memory/weak_ptr.h"
  34 #include "base/time/tick_clock.h"
  35 #include "base/time/time.h"
  36 #include "media/cast/common/transport_encryption_handler.h"
  37 #include "media/cast/logging/logging_defines.h"
  38 #include "media/cast/logging/simple_event_subscriber.h"
  39 #include "media/cast/net/cast_transport_config.h"
  40 #include "media/cast/net/cast_transport_sender.h"
  41 #include "media/cast/net/pacing/paced_sender.h"
  42 #include "media/cast/net/rtcp/rtcp.h"
  43 #include "media/cast/net/rtp/rtp_parser.h"
  44 #include "media/cast/net/rtp/rtp_sender.h"
  45
  46 namespace media {
  47 namespace cast {
  48
  49 class UdpTransport;
  50
  51 class CastTransportSenderImpl : public CastTransportSender {
  52  public:
  53   // |external_transport| is only used for testing.
  54   // |raw_events_callback|: Raw events will be returned on this callback
  55   // which will be invoked every |raw_events_callback_interval|.
  56   // This can be a null callback, i.e. if user is not interested in raw events.
  57   // |raw_events_callback_interval|: This can be |base::TimeDelta()| if
  58   // |raw_events_callback| is a null callback.
  59   // |options| contains optional settings for the transport, possible
  60   // keys are:
  61   //   "DSCP" (value ignored) - turns DSCP on
  62   //   "non_blocking_io" (value ignored) - Windows only.
  63   //                                       Turns on non-blocking IO for socket.
  64   //   "pacer_target_burst_size": int - specifies how many packets to send
  65   //                                    per 10 ms ideally.
  66   //   "pacer_max_burst_size": int - specifies how many pakcets to send
  67   //                                 per 10 ms, max
  68   //   "send_buffer_min_size": int - specifies the minimum socket send buffer
  69   //                                 size
  70   //   "disable_wifi_scan" (value ignored) - disable wifi scans while streaming
  71   //   "media_streaming_mode" (value ignored) - turn media streaming mode on
  72   // Note, these options may be ignored on some platforms.
  73   // TODO(hubbe): Too many callbacks, replace with an interface.
  74   CastTransportSenderImpl(
  75       net::NetLog* net_log,
  76       base::TickClock* clock,
  77       const net::IPEndPoint& local_end_point,
  78       const net::IPEndPoint& remote_end_point,
  79       scoped_ptr<base::DictionaryValue> options,
  80       const CastTransportStatusCallback& status_callback,
  81       const BulkRawEventsCallback& raw_events_callback,
  82       base::TimeDelta raw_events_callback_interval,
  83       const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner,
  84       const PacketReceiverCallback& packet_callback,
  85       PacketSender* external_transport);
  86
  87   ~CastTransportSenderImpl() override;
  88
  89   // CastTransportSender implementation.
  90   void InitializeAudio(const CastTransportRtpConfig& config,
  91                        const RtcpCastMessageCallback& cast_message_cb,
  92                        const RtcpRttCallback& rtt_cb) override;
  93   void InitializeVideo(const CastTransportRtpConfig& config,
  94                        const RtcpCastMessageCallback& cast_message_cb,
  95                        const RtcpRttCallback& rtt_cb) override;
  96   void InsertFrame(uint32 ssrc, const EncodedFrame& frame) override;
  97
  98   void SendSenderReport(uint32 ssrc,
  99                         base::TimeTicks current_time,
 100                         uint32 current_time_as_rtp_timestamp) override;
 101
 102   void CancelSendingFrames(uint32 ssrc,
 103                            const std::vector<uint32>& frame_ids) override;
 104
 105   void ResendFrameForKickstart(uint32 ssrc, uint32 frame_id) override;
 106
 107   PacketReceiverCallback PacketReceiverForTesting() override;
 108
 109   // CastTransportReceiver implementation.
 110   void AddValidSsrc(uint32 ssrc) override;
 111
 112   void SendRtcpFromRtpReceiver(
 113       uint32 ssrc,
 114       uint32 sender_ssrc,
 115       const RtcpTimeData& time_data,
 116       const RtcpCastMessage* cast_message,
 117       base::TimeDelta target_delay,
 118       const ReceiverRtcpEventSubscriber::RtcpEvents* rtcp_events,
 119       const RtpReceiverStatistics* rtp_receiver_statistics) override;
 120
 121  private:
 122   FRIEND_TEST_ALL_PREFIXES(CastTransportSenderImplTest, NacksCancelRetransmits);
 123   FRIEND_TEST_ALL_PREFIXES(CastTransportSenderImplTest, CancelRetransmits);
 124   FRIEND_TEST_ALL_PREFIXES(CastTransportSenderImplTest, Kickstart);
 125   FRIEND_TEST_ALL_PREFIXES(CastTransportSenderImplTest,
 126                            DedupRetransmissionWithAudio);
 127
 128   // Resend packets for the stream identified by |ssrc|.
 129   // If |cancel_rtx_if_not_in_list| is true then transmission of packets for the
 130   // frames but not in the list will be dropped.
 131   // See PacedSender::ResendPackets() to see how |dedup_info| works.
 132   void ResendPackets(uint32 ssrc,
 133                      const MissingFramesAndPacketsMap& missing_packets,
 134                      bool cancel_rtx_if_not_in_list,
 135                      const DedupInfo& dedup_info);
 136
 137   // If |raw_events_callback_| is non-null, calls it with events collected
 138   // by |event_subscriber_| since last call.
 139   void SendRawEvents();
 140
 141   // Called when a packet is received.
 142   bool OnReceivedPacket(scoped_ptr<Packet> packet);
 143
 144   // Called when a log message is received.
 145   void OnReceivedLogMessage(EventMediaType media_type,
 146                             const RtcpReceiverLogMessage& log);
 147
 148   // Called when a RTCP Cast message is received.
 149   void OnReceivedCastMessage(uint32 ssrc,
 150                              const RtcpCastMessageCallback& cast_message_cb,
 151                              const RtcpCastMessage& cast_message);
 152
 153   base::TickClock* clock_;  // Not owned by this class.
 154   CastTransportStatusCallback status_callback_;
 155   scoped_refptr<base::SingleThreadTaskRunner> transport_task_runner_;
 156
 157   LoggingImpl logging_;
 158
 159   // Interface to a UDP socket.
 160   scoped_ptr<UdpTransport> transport_;
 161
 162   // Packet sender that performs pacing.
 163   PacedSender pacer_;
 164
 165   // Packetizer for audio and video frames.
 166   scoped_ptr<RtpSender> audio_sender_;
 167   scoped_ptr<RtpSender> video_sender_;
 168
 169   // Maintains RTCP session for audio and video.
 170   scoped_ptr<Rtcp> audio_rtcp_session_;
 171   scoped_ptr<Rtcp> video_rtcp_session_;
 172
 173   // Encrypts data in EncodedFrames before they are sent.  Note that it's
 174   // important for the encryption to happen here, in code that would execute in
 175   // the main browser process, for security reasons.  This helps to mitigate
 176   // the damage that could be caused by a compromised renderer process.
 177   TransportEncryptionHandler audio_encryptor_;
 178   TransportEncryptionHandler video_encryptor_;
 179
 180   // This is non-null iff |raw_events_callback_| is non-null.
 181   scoped_ptr<SimpleEventSubscriber> event_subscriber_;
 182
 183   BulkRawEventsCallback raw_events_callback_;
 184   base::TimeDelta raw_events_callback_interval_;
 185
 186   // Right after a frame is sent we record the number of bytes sent to the
 187   // socket. We record the corresponding bytes sent for the most recent ACKed
 188   // audio packet.
 189   int64 last_byte_acked_for_audio_;
 190
 191   // Packets that don't match these ssrcs are ignored.
 192   std::set<uint32> valid_ssrcs_;
 193
 194   // Called with incoming packets. (Unless they match the
 195   // channels created by Initialize{Audio,Video}.
 196   PacketReceiverCallback packet_callback_;
 197
 198   scoped_ptr<net::ScopedWifiOptions> wifi_options_autoreset_;
 199
 200   base::WeakPtrFactory<CastTransportSenderImpl> weak_factory_;
 201
 202   DISALLOW_COPY_AND_ASSIGN(CastTransportSenderImpl);
 203 };
 204
 205 }  // namespace cast
 206 }  // namespace media
 207
 208 #endif  // MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_IMPL_H_