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