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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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| 32 #include "webrtc/media/base/videosourceinterface.h" | 32 #include "webrtc/media/base/videosourceinterface.h" |
| 33 #include "webrtc/p2p/base/transportcontroller.h" | 33 #include "webrtc/p2p/base/transportcontroller.h" |
| 34 #include "webrtc/p2p/client/socketmonitor.h" | 34 #include "webrtc/p2p/client/socketmonitor.h" |
| 35 #include "webrtc/pc/audiomonitor.h" | 35 #include "webrtc/pc/audiomonitor.h" |
| 36 #include "webrtc/pc/bundlefilter.h" | 36 #include "webrtc/pc/bundlefilter.h" |
| 37 #include "webrtc/pc/mediamonitor.h" | 37 #include "webrtc/pc/mediamonitor.h" |
| 38 #include "webrtc/pc/mediasession.h" | 38 #include "webrtc/pc/mediasession.h" |
| 39 #include "webrtc/pc/rtcpmuxfilter.h" | 39 #include "webrtc/pc/rtcpmuxfilter.h" |
| 40 #include "webrtc/pc/srtpfilter.h" | 40 #include "webrtc/pc/srtpfilter.h" |
| 41 | 41 |
| 42 namespace rtc { | |
| 43 struct PacketTransport; | |
| 44 } | |
| 45 | |
| 42 namespace webrtc { | 46 namespace webrtc { |
| 43 class AudioSinkInterface; | 47 class AudioSinkInterface; |
| 44 } // namespace webrtc | 48 } // namespace webrtc |
| 45 | 49 |
| 46 namespace cricket { | 50 namespace cricket { |
| 47 | 51 |
| 48 struct CryptoParams; | 52 struct CryptoParams; |
| 49 class MediaContentDescription; | 53 class MediaContentDescription; |
| 50 | 54 |
| 51 // BaseChannel contains logic common to voice and video, including enable, | 55 // BaseChannel contains logic common to voice and video, including enable, |
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| 226 | 230 |
| 227 void FlushRtcpMessages_n(); | 231 void FlushRtcpMessages_n(); |
| 228 | 232 |
| 229 // NetworkInterface implementation, called by MediaEngine | 233 // NetworkInterface implementation, called by MediaEngine |
| 230 bool SendPacket(rtc::CopyOnWriteBuffer* packet, | 234 bool SendPacket(rtc::CopyOnWriteBuffer* packet, |
| 231 const rtc::PacketOptions& options) override; | 235 const rtc::PacketOptions& options) override; |
| 232 bool SendRtcp(rtc::CopyOnWriteBuffer* packet, | 236 bool SendRtcp(rtc::CopyOnWriteBuffer* packet, |
| 233 const rtc::PacketOptions& options) override; | 237 const rtc::PacketOptions& options) override; |
| 234 | 238 |
| 235 // From TransportChannel | 239 // From TransportChannel |
| 236 void OnWritableState(TransportChannel* channel); | 240 void OnWritableState(rtc::PacketTransport* pt); |
| 237 virtual void OnChannelRead(TransportChannel* channel, | 241 virtual void OnChannelRead(rtc::PacketTransport* pt, |
| 238 const char* data, | 242 const char* data, |
| 239 size_t len, | 243 size_t len, |
| 240 const rtc::PacketTime& packet_time, | 244 const rtc::PacketTime& packet_time, |
| 241 int flags); | 245 int flags); |
|
pthatcher1
2016/10/13 20:26:38
OnChannelRead should probably be renamed to OnPack
johan
2016/10/14 16:09:33
Acknowledged.
| |
| 242 void OnReadyToSend(TransportChannel* channel); | 246 void OnReadyToSend(rtc::PacketTransport* pt); |
| 243 | 247 |
| 244 void OnDtlsState(TransportChannel* channel, DtlsTransportState state); | 248 void OnDtlsState(TransportChannel* channel, DtlsTransportState state); |
| 245 | 249 |
| 246 void OnSelectedCandidatePairChanged( | 250 void OnSelectedCandidatePairChanged( |
| 247 TransportChannel* channel, | 251 TransportChannel* channel, |
| 248 CandidatePairInterface* selected_candidate_pair, | 252 CandidatePairInterface* selected_candidate_pair, |
| 249 int last_sent_packet_id, | 253 int last_sent_packet_id, |
| 250 bool ready_to_send); | 254 bool ready_to_send); |
| 251 | 255 |
| 252 bool PacketIsRtcp(const TransportChannel* channel, const char* data, | 256 bool PacketIsRtcp(const TransportChannel* channel, const char* data, |
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| 349 template <class FunctorT> | 353 template <class FunctorT> |
| 350 bool InvokeOnWorker(const rtc::Location& posted_from, | 354 bool InvokeOnWorker(const rtc::Location& posted_from, |
| 351 const FunctorT& functor) { | 355 const FunctorT& functor) { |
| 352 return worker_thread_->Invoke<bool>(posted_from, functor); | 356 return worker_thread_->Invoke<bool>(posted_from, functor); |
| 353 } | 357 } |
| 354 | 358 |
| 355 private: | 359 private: |
| 356 bool InitNetwork_n(const std::string* bundle_transport_name); | 360 bool InitNetwork_n(const std::string* bundle_transport_name); |
| 357 void DisconnectTransportChannels_n(); | 361 void DisconnectTransportChannels_n(); |
| 358 void DestroyTransportChannels_n(); | 362 void DestroyTransportChannels_n(); |
| 359 void SignalSentPacket_n(TransportChannel* channel, | 363 void SignalSentPacket_n(rtc::PacketTransport* pt, |
| 360 const rtc::SentPacket& sent_packet); | 364 const rtc::SentPacket& sent_packet); |
| 361 void SignalSentPacket_w(const rtc::SentPacket& sent_packet); | 365 void SignalSentPacket_w(const rtc::SentPacket& sent_packet); |
| 362 bool IsReadyToSendMedia_n() const; | 366 bool IsReadyToSendMedia_n() const; |
| 363 void CacheRtpAbsSendTimeHeaderExtension_n(int rtp_abs_sendtime_extn_id); | 367 void CacheRtpAbsSendTimeHeaderExtension_n(int rtp_abs_sendtime_extn_id); |
| 364 | 368 |
| 365 rtc::Thread* const worker_thread_; | 369 rtc::Thread* const worker_thread_; |
| 366 rtc::Thread* const network_thread_; | 370 rtc::Thread* const network_thread_; |
| 367 rtc::AsyncInvoker invoker_; | 371 rtc::AsyncInvoker invoker_; |
| 368 | 372 |
| 369 const std::string content_name_; | 373 const std::string content_name_; |
| 370 std::unique_ptr<ConnectionMonitor> connection_monitor_; | 374 std::unique_ptr<ConnectionMonitor> connection_monitor_; |
| 371 | 375 |
| 372 // Transport related members that should be accessed from network thread. | 376 // Transport related members that should be accessed from network thread. |
| 373 TransportController* const transport_controller_; | 377 TransportController* const transport_controller_; |
| 374 std::string transport_name_; | 378 std::string transport_name_; |
| 375 // Is RTCP used at all by this type of channel? | 379 // Is RTCP used at all by this type of channel? |
| 376 // Expected to be true (as of typing this) for everything except data | 380 // Expected to be true (as of typing this) for everything except data |
| 377 // channels. | 381 // channels. |
| 378 const bool rtcp_enabled_; | 382 const bool rtcp_enabled_; |
| 383 // TODO(johan): replace TransportChannel* by rtc::PacketTransport*. | |
| 379 TransportChannel* transport_channel_ = nullptr; | 384 TransportChannel* transport_channel_ = nullptr; |
| 380 std::vector<std::pair<rtc::Socket::Option, int> > socket_options_; | 385 std::vector<std::pair<rtc::Socket::Option, int> > socket_options_; |
| 381 TransportChannel* rtcp_transport_channel_ = nullptr; | 386 TransportChannel* rtcp_transport_channel_ = nullptr; |
| 382 std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_; | 387 std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_; |
| 383 SrtpFilter srtp_filter_; | 388 SrtpFilter srtp_filter_; |
| 384 RtcpMuxFilter rtcp_mux_filter_; | 389 RtcpMuxFilter rtcp_mux_filter_; |
| 385 BundleFilter bundle_filter_; | 390 BundleFilter bundle_filter_; |
| 386 bool rtp_ready_to_send_ = false; | 391 bool rtp_ready_to_send_ = false; |
| 387 bool rtcp_ready_to_send_ = false; | 392 bool rtcp_ready_to_send_ = false; |
| 388 bool writable_ = false; | 393 bool writable_ = false; |
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| 477 void GetActiveStreams_w(AudioInfo::StreamList* actives); | 482 void GetActiveStreams_w(AudioInfo::StreamList* actives); |
| 478 webrtc::RtpParameters GetRtpSendParameters_w(uint32_t ssrc) const; | 483 webrtc::RtpParameters GetRtpSendParameters_w(uint32_t ssrc) const; |
| 479 bool SetRtpSendParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters); | 484 bool SetRtpSendParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters); |
| 480 webrtc::RtpParameters GetRtpReceiveParameters_w(uint32_t ssrc) const; | 485 webrtc::RtpParameters GetRtpReceiveParameters_w(uint32_t ssrc) const; |
| 481 bool SetRtpReceiveParameters_w(uint32_t ssrc, | 486 bool SetRtpReceiveParameters_w(uint32_t ssrc, |
| 482 webrtc::RtpParameters parameters); | 487 webrtc::RtpParameters parameters); |
| 483 cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_AUDIO; } | 488 cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_AUDIO; } |
| 484 | 489 |
| 485 private: | 490 private: |
| 486 // overrides from BaseChannel | 491 // overrides from BaseChannel |
| 487 void OnChannelRead(TransportChannel* channel, | 492 void OnChannelRead(rtc::PacketTransport* pt, |
| 488 const char* data, | 493 const char* data, |
| 489 size_t len, | 494 size_t len, |
| 490 const rtc::PacketTime& packet_time, | 495 const rtc::PacketTime& packet_time, |
| 491 int flags) override; | 496 int flags) override; |
| 492 void UpdateMediaSendRecvState_w() override; | 497 void UpdateMediaSendRecvState_w() override; |
| 493 const ContentInfo* GetFirstContent(const SessionDescription* sdesc) override; | 498 const ContentInfo* GetFirstContent(const SessionDescription* sdesc) override; |
| 494 bool SetLocalContent_w(const MediaContentDescription* content, | 499 bool SetLocalContent_w(const MediaContentDescription* content, |
| 495 ContentAction action, | 500 ContentAction action, |
| 496 std::string* error_desc) override; | 501 std::string* error_desc) override; |
| 497 bool SetRemoteContent_w(const MediaContentDescription* content, | 502 bool SetRemoteContent_w(const MediaContentDescription* content, |
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| 721 // SetSendParameters. | 726 // SetSendParameters. |
| 722 DataSendParameters last_send_params_; | 727 DataSendParameters last_send_params_; |
| 723 // Last DataRecvParameters sent down to the media_channel() via | 728 // Last DataRecvParameters sent down to the media_channel() via |
| 724 // SetRecvParameters. | 729 // SetRecvParameters. |
| 725 DataRecvParameters last_recv_params_; | 730 DataRecvParameters last_recv_params_; |
| 726 }; | 731 }; |
| 727 | 732 |
| 728 } // namespace cricket | 733 } // namespace cricket |
| 729 | 734 |
| 730 #endif // WEBRTC_PC_CHANNEL_H_ | 735 #endif // WEBRTC_PC_CHANNEL_H_ |
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