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1 /* | 1 /* |
2 * Copyright 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2004 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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26 #include "webrtc/media/base/videosinkinterface.h" | 26 #include "webrtc/media/base/videosinkinterface.h" |
27 #include "webrtc/media/base/videosourceinterface.h" | 27 #include "webrtc/media/base/videosourceinterface.h" |
28 #include "webrtc/p2p/base/dtlstransportinternal.h" | 28 #include "webrtc/p2p/base/dtlstransportinternal.h" |
29 #include "webrtc/p2p/base/packettransportinternal.h" | 29 #include "webrtc/p2p/base/packettransportinternal.h" |
30 #include "webrtc/p2p/base/transportcontroller.h" | 30 #include "webrtc/p2p/base/transportcontroller.h" |
31 #include "webrtc/p2p/client/socketmonitor.h" | 31 #include "webrtc/p2p/client/socketmonitor.h" |
32 #include "webrtc/pc/audiomonitor.h" | 32 #include "webrtc/pc/audiomonitor.h" |
33 #include "webrtc/pc/mediamonitor.h" | 33 #include "webrtc/pc/mediamonitor.h" |
34 #include "webrtc/pc/mediasession.h" | 34 #include "webrtc/pc/mediasession.h" |
35 #include "webrtc/pc/rtcpmuxfilter.h" | 35 #include "webrtc/pc/rtcpmuxfilter.h" |
36 #include "webrtc/pc/rtptransportinternal.h" | |
37 #include "webrtc/pc/srtpfilter.h" | 36 #include "webrtc/pc/srtpfilter.h" |
38 #include "webrtc/rtc_base/asyncinvoker.h" | 37 #include "webrtc/rtc_base/asyncinvoker.h" |
39 #include "webrtc/rtc_base/asyncudpsocket.h" | 38 #include "webrtc/rtc_base/asyncudpsocket.h" |
40 #include "webrtc/rtc_base/criticalsection.h" | 39 #include "webrtc/rtc_base/criticalsection.h" |
41 #include "webrtc/rtc_base/network.h" | 40 #include "webrtc/rtc_base/network.h" |
42 #include "webrtc/rtc_base/sigslot.h" | 41 #include "webrtc/rtc_base/sigslot.h" |
43 #include "webrtc/rtc_base/window.h" | 42 #include "webrtc/rtc_base/window.h" |
44 | 43 |
45 namespace webrtc { | 44 namespace webrtc { |
46 class AudioSinkInterface; | 45 class AudioSinkInterface; |
46 class RtpTransportInternal; | |
47 class SrtpTransport; | |
47 } // namespace webrtc | 48 } // namespace webrtc |
48 | 49 |
49 namespace cricket { | 50 namespace cricket { |
50 | 51 |
51 struct CryptoParams; | 52 struct CryptoParams; |
52 class MediaContentDescription; | 53 class MediaContentDescription; |
53 | 54 |
54 // BaseChannel contains logic common to voice and video, including enable, | 55 // BaseChannel contains logic common to voice and video, including enable, |
55 // marshaling calls to a worker and network threads, and connection and media | 56 // marshaling calls to a worker and network threads, and connection and media |
56 // monitors. | 57 // monitors. |
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92 // done. | 93 // done. |
93 void Deinit(); | 94 void Deinit(); |
94 | 95 |
95 rtc::Thread* worker_thread() const { return worker_thread_; } | 96 rtc::Thread* worker_thread() const { return worker_thread_; } |
96 rtc::Thread* network_thread() const { return network_thread_; } | 97 rtc::Thread* network_thread() const { return network_thread_; } |
97 const std::string& content_name() const { return content_name_; } | 98 const std::string& content_name() const { return content_name_; } |
98 // TODO(deadbeef): This is redundant; remove this. | 99 // TODO(deadbeef): This is redundant; remove this. |
99 const std::string& transport_name() const { return transport_name_; } | 100 const std::string& transport_name() const { return transport_name_; } |
100 bool enabled() const { return enabled_; } | 101 bool enabled() const { return enabled_; } |
101 | 102 |
102 // This function returns true if we are using SRTP. | 103 // This function returns true if we are using SRTP. |
Taylor Brandstetter
2017/08/26 02:40:39
nit: Should update this comment.
Zhi Huang
2017/08/29 18:40:34
Done.
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103 bool secure() const { return srtp_filter_.IsActive(); } | 104 bool secure_sdes() const { return srtp_filter_.IsActive(); } |
pthatcher
2017/08/28 21:42:56
While we're changing the method name, can we make
Zhi Huang
2017/08/29 18:40:34
Done.
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104 // The following function returns true if we are using | 105 // The following function returns true if we are using |
105 // DTLS-based keying. If you turned off SRTP later, however | 106 // DTLS-based keying. If you turned off SRTP later, however |
106 // you could have secure() == false and dtls_secure() == true. | 107 // you could have secure() == false and dtls_secure() == true. |
107 bool secure_dtls() const { return dtls_keyed_; } | 108 bool secure_dtls() const { return dtls_active_; } |
109 | |
110 bool secure() const { return secure_sdes() || secure_dtls(); } | |
pthatcher
2017/08/28 21:42:56
Would "srtp_active" be a good name here, to along
Zhi Huang
2017/08/29 18:40:34
It sounds good to me.
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108 | 111 |
109 bool writable() const { return writable_; } | 112 bool writable() const { return writable_; } |
110 | 113 |
111 // Set the transport(s), and update writability and "ready-to-send" state. | 114 // Set the transport(s), and update writability and "ready-to-send" state. |
112 // |rtp_transport| must be non-null. | 115 // |rtp_transport| must be non-null. |
113 // |rtcp_transport| must be supplied if NeedsRtcpTransport() is true (meaning | 116 // |rtcp_transport| must be supplied if NeedsRtcpTransport() is true (meaning |
114 // RTCP muxing is not fully active yet). | 117 // RTCP muxing is not fully active yet). |
115 // |rtp_transport| and |rtcp_transport| must share the same transport name as | 118 // |rtp_transport| and |rtcp_transport| must share the same transport name as |
116 // well. | 119 // well. |
117 // Can not start with "rtc::PacketTransportInternal" and switch to | 120 // Can not start with "rtc::PacketTransportInternal" and switch to |
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391 std::string transport_name_; | 394 std::string transport_name_; |
392 | 395 |
393 const bool rtcp_mux_required_; | 396 const bool rtcp_mux_required_; |
394 | 397 |
395 // Separate DTLS/non-DTLS pointers to support using BaseChannel without DTLS. | 398 // Separate DTLS/non-DTLS pointers to support using BaseChannel without DTLS. |
396 // Temporary measure until more refactoring is done. | 399 // Temporary measure until more refactoring is done. |
397 // If non-null, "X_dtls_transport_" will always equal "X_packet_transport_". | 400 // If non-null, "X_dtls_transport_" will always equal "X_packet_transport_". |
398 DtlsTransportInternal* rtp_dtls_transport_ = nullptr; | 401 DtlsTransportInternal* rtp_dtls_transport_ = nullptr; |
399 DtlsTransportInternal* rtcp_dtls_transport_ = nullptr; | 402 DtlsTransportInternal* rtcp_dtls_transport_ = nullptr; |
400 std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport_; | 403 std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport_; |
404 webrtc::SrtpTransport* srtp_transport_ = nullptr; | |
401 std::vector<std::pair<rtc::Socket::Option, int> > socket_options_; | 405 std::vector<std::pair<rtc::Socket::Option, int> > socket_options_; |
402 std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_; | 406 std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_; |
403 SrtpFilter srtp_filter_; | 407 SrtpFilter srtp_filter_; |
pthatcher
2017/08/28 21:42:56
It feels like this really should be renamed to Sde
Zhi Huang
2017/08/29 18:40:34
What about sdes_negotiator_? I plan to rename the
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404 RtcpMuxFilter rtcp_mux_filter_; | 408 RtcpMuxFilter rtcp_mux_filter_; |
405 bool writable_ = false; | 409 bool writable_ = false; |
406 bool was_ever_writable_ = false; | 410 bool was_ever_writable_ = false; |
407 bool has_received_packet_ = false; | 411 bool has_received_packet_ = false; |
408 bool dtls_keyed_ = false; | 412 bool dtls_active_ = false; |
409 const bool srtp_required_ = true; | 413 const bool srtp_required_ = true; |
410 int rtp_abs_sendtime_extn_id_ = -1; | |
411 | 414 |
412 // MediaChannel related members that should be accessed from the worker | 415 // MediaChannel related members that should be accessed from the worker |
413 // thread. | 416 // thread. |
414 MediaChannel* const media_channel_; | 417 MediaChannel* const media_channel_; |
415 // Currently the |enabled_| flag is accessed from the signaling thread as | 418 // Currently the |enabled_| flag is accessed from the signaling thread as |
416 // well, but it can be changed only when signaling thread does a synchronous | 419 // well, but it can be changed only when signaling thread does a synchronous |
417 // call to the worker thread, so it should be safe. | 420 // call to the worker thread, so it should be safe. |
418 bool enabled_ = false; | 421 bool enabled_ = false; |
419 std::vector<StreamParams> local_streams_; | 422 std::vector<StreamParams> local_streams_; |
420 std::vector<StreamParams> remote_streams_; | 423 std::vector<StreamParams> remote_streams_; |
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728 // SetSendParameters. | 731 // SetSendParameters. |
729 DataSendParameters last_send_params_; | 732 DataSendParameters last_send_params_; |
730 // Last DataRecvParameters sent down to the media_channel() via | 733 // Last DataRecvParameters sent down to the media_channel() via |
731 // SetRecvParameters. | 734 // SetRecvParameters. |
732 DataRecvParameters last_recv_params_; | 735 DataRecvParameters last_recv_params_; |
733 }; | 736 }; |
734 | 737 |
735 } // namespace cricket | 738 } // namespace cricket |
736 | 739 |
737 #endif // WEBRTC_PC_CHANNEL_H_ | 740 #endif // WEBRTC_PC_CHANNEL_H_ |
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