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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 1 // Copyright 2013 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 // This is the main interface for the cast transport sender. It accepts encoded | 5 // This is the main interface for the cast transport sender. It accepts encoded |
6 // frames (both audio and video), encrypts their encoded data, packetizes them | 6 // frames (both audio and video), encrypts their encoded data, packetizes them |
7 // and feeds them into a transport (e.g., UDP). | 7 // and feeds them into a transport (e.g., UDP). |
8 | 8 |
9 // Construction of the Cast Sender and the Cast Transport Sender should be done | 9 // Construction of the Cast Sender and the Cast Transport Sender should be done |
10 // in the following order: | 10 // in the following order: |
11 // 1. Create CastTransportSender. | 11 // 1. Create CastTransportSender. |
12 // 2. Create CastSender (accepts CastTransportSender as an input). | 12 // 2. Create CastSender (accepts CastTransportSender as an input). |
13 // 3. Call CastTransportSender::SetPacketReceiver to ensure that the packets | |
14 // received by the CastTransportSender will be sent to the CastSender. | |
15 // Steps 3 can be done interchangeably. | |
16 | 13 |
17 // Destruction: The CastTransportSender is assumed to be valid as long as the | 14 // Destruction: The CastTransportSender is assumed to be valid as long as the |
18 // CastSender is alive. Therefore the CastSender should be destructed before the | 15 // CastSender is alive. Therefore the CastSender should be destructed before the |
19 // CastTransportSender. | 16 // CastTransportSender. |
20 // This also works when the CastSender acts as a receiver for the RTCP packets | |
21 // due to the weak pointers in the ReceivedPacket method in cast_sender_impl.cc. | |
22 | 17 |
23 #ifndef MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_ | 18 #ifndef MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_ |
24 #define MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_ | 19 #define MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_ |
25 | 20 |
26 #include "base/basictypes.h" | 21 #include "base/basictypes.h" |
27 #include "base/callback.h" | 22 #include "base/callback.h" |
28 #include "base/single_thread_task_runner.h" | 23 #include "base/single_thread_task_runner.h" |
29 #include "base/threading/non_thread_safe.h" | 24 #include "base/threading/non_thread_safe.h" |
30 #include "base/time/tick_clock.h" | 25 #include "base/time/tick_clock.h" |
31 #include "media/cast/logging/logging_defines.h" | 26 #include "media/cast/logging/logging_defines.h" |
32 #include "media/cast/net/cast_transport_config.h" | 27 #include "media/cast/net/cast_transport_config.h" |
33 #include "media/cast/net/cast_transport_defines.h" | 28 #include "media/cast/net/cast_transport_defines.h" |
29 #include "media/cast/net/rtcp/receiver_rtcp_event_subscriber.h" | |
30 #include "media/cast/net/rtcp/rtcp_defines.h" | |
34 | 31 |
35 namespace net { | 32 namespace net { |
36 class NetLog; | 33 class NetLog; |
37 } // namespace net | 34 } // namespace net |
38 | 35 |
39 namespace media { | 36 namespace media { |
40 namespace cast { | 37 namespace cast { |
41 | 38 |
42 // Following the initialization of either audio or video an initialization | 39 // Following the initialization of either audio or video an initialization |
43 // status will be sent via this callback. | 40 // status will be sent via this callback. |
44 typedef base::Callback<void(CastTransportStatus status)> | 41 typedef base::Callback<void(CastTransportStatus status)> |
45 CastTransportStatusCallback; | 42 CastTransportStatusCallback; |
46 | 43 |
47 typedef base::Callback<void(const std::vector<PacketEvent>&)> | 44 typedef base::Callback<void(const std::vector<PacketEvent>&, |
45 const std::vector<FrameEvent>&)> | |
48 BulkRawEventsCallback; | 46 BulkRawEventsCallback; |
49 | 47 |
50 // The application should only trigger this class from the transport thread. | 48 // The application should only trigger this class from the transport thread. |
51 class CastTransportSender : public base::NonThreadSafe { | 49 class CastTransportSender : public base::NonThreadSafe { |
52 public: | 50 public: |
53 static scoped_ptr<CastTransportSender> Create( | 51 static scoped_ptr<CastTransportSender> Create( |
54 net::NetLog* net_log, | 52 net::NetLog* net_log, |
55 base::TickClock* clock, | 53 base::TickClock* clock, |
56 const net::IPEndPoint& remote_end_point, | 54 const net::IPEndPoint& remote_end_point, |
57 const CastTransportStatusCallback& status_callback, | 55 const CastTransportStatusCallback& status_callback, |
58 const BulkRawEventsCallback& raw_events_callback, | 56 const BulkRawEventsCallback& raw_events_callback, |
59 base::TimeDelta raw_events_callback_interval, | 57 base::TimeDelta raw_events_callback_interval, |
60 const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner); | 58 const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner); |
61 | 59 |
62 virtual ~CastTransportSender() {} | 60 virtual ~CastTransportSender() {} |
63 | 61 |
64 // Audio/Video initialization. | 62 // Audio/Video initialization. |
65 // Encoded frames cannot be transmitted until the relevant initialize method | 63 // Encoded frames cannot be transmitted until the relevant initialize method |
66 // is called. Usually called by CastSender. | 64 // is called. |
67 virtual void InitializeAudio(const CastTransportRtpConfig& config) = 0; | 65 virtual void InitializeAudio(const CastTransportRtpConfig& config, |
68 virtual void InitializeVideo(const CastTransportRtpConfig& config) = 0; | 66 const RtcpCastMessageCallback& cast_message_cb, |
69 | 67 const RtcpRttCallback& rtt_cb) = 0; |
70 // Sets the Cast packet receiver. Should be called after creation on the | 68 virtual void InitializeVideo(const CastTransportRtpConfig& config, |
71 // Cast sender. Packets won't be received until this function is called. | 69 const RtcpCastMessageCallback& cast_message_cb, |
72 virtual void SetPacketReceiver( | 70 const RtcpRttCallback& rtt_cb) = 0; |
73 const PacketReceiverCallback& packet_receiver) = 0; | |
74 | 71 |
75 // The following two functions handle the encoded media frames (audio and | 72 // The following two functions handle the encoded media frames (audio and |
76 // video) to be processed. | 73 // video) to be processed. |
77 // Frames will be encrypted, packetized and transmitted to the network. | 74 // Frames will be encrypted, packetized and transmitted to the network. |
78 virtual void InsertCodedAudioFrame(const EncodedFrame& audio_frame) = 0; | 75 virtual void InsertCodedAudioFrame(const EncodedFrame& audio_frame) = 0; |
79 virtual void InsertCodedVideoFrame(const EncodedFrame& video_frame) = 0; | 76 virtual void InsertCodedVideoFrame(const EncodedFrame& video_frame) = 0; |
80 | 77 |
81 // Builds an RTCP packet and sends it to the network. | 78 // Sends a RTCP sender report to the receiver. |
82 // |ntp_seconds|, |ntp_fraction| and |rtp_timestamp| are used in the | 79 // |audio| is true of this is for audio stream. False otherwise. |
83 // RTCP Sender Report. | 80 // |current_time| is the current time reported by a tick clock. |
84 virtual void SendRtcpFromRtpSender(uint32 packet_type_flags, | 81 // |current_time_as_rtp_timestamp| is the corresponding RTP timestamp. |
85 uint32 ntp_seconds, | 82 virtual void SendRtcpFromRtpSender( |
miu
2014/07/16 00:09:30
Can we name this SendSenderReport instead? Or, ev
Alpha Left Google
2014/07/17 01:01:43
Let's use the RTCP terminology of "SenderReport".
| |
86 uint32 ntp_fraction, | 83 bool audio, |
87 uint32 rtp_timestamp, | 84 base::TimeTicks current_time, |
88 const RtcpDlrrReportBlock& dlrr, | 85 uint32 current_time_as_rtp_timestamp) = 0; |
89 uint32 sending_ssrc, | |
90 const std::string& c_name) = 0; | |
91 | 86 |
92 // Retransmission request. | 87 // Retransmission request. |
93 // |missing_packets| includes the list of frames and packets in each | 88 // |missing_packets| includes the list of frames and packets in each |
94 // frame to be re-transmitted. | 89 // frame to be re-transmitted. |
95 // If |cancel_rtx_if_not_in_list| is used as an optimization to cancel | 90 // If |cancel_rtx_if_not_in_list| is used as an optimization to cancel |
96 // pending re-transmission requests of packets not listed in | 91 // pending re-transmission requests of packets not listed in |
97 // |missing_packets|. If the requested packet(s) were sent recently | 92 // |missing_packets|. If the requested packet(s) were sent recently |
98 // (how long is specified by |dedupe_window|) then this re-transmit | 93 // (how long is specified by |dedupe_window|) then this re-transmit |
99 // will be ignored. | 94 // will be ignored. |
100 virtual void ResendPackets( | 95 virtual void ResendPackets( |
101 bool is_audio, | 96 bool is_audio, |
102 const MissingFramesAndPacketsMap& missing_packets, | 97 const MissingFramesAndPacketsMap& missing_packets, |
103 bool cancel_rtx_if_not_in_list, | 98 bool cancel_rtx_if_not_in_list, |
104 base::TimeDelta dedupe_window) = 0; | 99 base::TimeDelta dedupe_window) = 0; |
100 | |
101 // Returns a callback for receiving packets for testing purposes. | |
102 virtual PacketReceiverCallback PacketReceiverForTesting() = 0; | |
105 }; | 103 }; |
106 | 104 |
107 } // namespace cast | 105 } // namespace cast |
108 } // namespace media | 106 } // namespace media |
109 | 107 |
110 #endif // MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_ | 108 #endif // MEDIA_CAST_NET_CAST_TRANSPORT_SENDER_H_ |
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