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1 /* | 1 /* |
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 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 |
11 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 11 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 #include <string> | 15 #include <string> |
16 #include <vector> | 16 #include <vector> |
17 | 17 |
18 #include "webrtc/api/rtpparameters.h" | 18 #include "webrtc/api/rtpparameters.h" |
19 #include "webrtc/base/basictypes.h" | 19 #include "webrtc/base/basictypes.h" |
20 #include "webrtc/base/buffer.h" | 20 #include "webrtc/base/copyonwritebuffer.h" |
21 #include "webrtc/base/dscp.h" | 21 #include "webrtc/base/dscp.h" |
22 #include "webrtc/base/logging.h" | 22 #include "webrtc/base/logging.h" |
23 #include "webrtc/base/optional.h" | 23 #include "webrtc/base/optional.h" |
24 #include "webrtc/base/sigslot.h" | 24 #include "webrtc/base/sigslot.h" |
25 #include "webrtc/base/socket.h" | 25 #include "webrtc/base/socket.h" |
26 #include "webrtc/base/window.h" | 26 #include "webrtc/base/window.h" |
27 #include "webrtc/media/base/codec.h" | 27 #include "webrtc/media/base/codec.h" |
28 #include "webrtc/media/base/mediaconstants.h" | 28 #include "webrtc/media/base/mediaconstants.h" |
29 #include "webrtc/media/base/streamparams.h" | 29 #include "webrtc/media/base/streamparams.h" |
30 #include "webrtc/media/base/videosinkinterface.h" | 30 #include "webrtc/media/base/videosinkinterface.h" |
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349 return &(*it); | 349 return &(*it); |
350 } | 350 } |
351 return NULL; | 351 return NULL; |
352 } | 352 } |
353 | 353 |
354 class MediaChannel : public sigslot::has_slots<> { | 354 class MediaChannel : public sigslot::has_slots<> { |
355 public: | 355 public: |
356 class NetworkInterface { | 356 class NetworkInterface { |
357 public: | 357 public: |
358 enum SocketType { ST_RTP, ST_RTCP }; | 358 enum SocketType { ST_RTP, ST_RTCP }; |
359 virtual bool SendPacket(rtc::Buffer* packet, | 359 virtual bool SendPacket(rtc::CopyOnWriteBuffer* packet, |
360 const rtc::PacketOptions& options) = 0; | 360 const rtc::PacketOptions& options) = 0; |
361 virtual bool SendRtcp(rtc::Buffer* packet, | 361 virtual bool SendRtcp(rtc::CopyOnWriteBuffer* packet, |
362 const rtc::PacketOptions& options) = 0; | 362 const rtc::PacketOptions& options) = 0; |
363 virtual int SetOption(SocketType type, rtc::Socket::Option opt, | 363 virtual int SetOption(SocketType type, rtc::Socket::Option opt, |
364 int option) = 0; | 364 int option) = 0; |
365 virtual ~NetworkInterface() {} | 365 virtual ~NetworkInterface() {} |
366 }; | 366 }; |
367 | 367 |
368 MediaChannel(const MediaConfig& config) | 368 MediaChannel(const MediaConfig& config) |
369 : enable_dscp_(config.enable_dscp), network_interface_(NULL) {} | 369 : enable_dscp_(config.enable_dscp), network_interface_(NULL) {} |
370 MediaChannel() : enable_dscp_(false), network_interface_(NULL) {} | 370 MediaChannel() : enable_dscp_(false), network_interface_(NULL) {} |
371 virtual ~MediaChannel() {} | 371 virtual ~MediaChannel() {} |
372 | 372 |
373 // Sets the abstract interface class for sending RTP/RTCP data. | 373 // Sets the abstract interface class for sending RTP/RTCP data. |
374 virtual void SetInterface(NetworkInterface *iface) { | 374 virtual void SetInterface(NetworkInterface *iface) { |
375 rtc::CritScope cs(&network_interface_crit_); | 375 rtc::CritScope cs(&network_interface_crit_); |
376 network_interface_ = iface; | 376 network_interface_ = iface; |
377 SetDscp(enable_dscp_ ? PreferredDscp() : rtc::DSCP_DEFAULT); | 377 SetDscp(enable_dscp_ ? PreferredDscp() : rtc::DSCP_DEFAULT); |
378 } | 378 } |
379 virtual rtc::DiffServCodePoint PreferredDscp() const { | 379 virtual rtc::DiffServCodePoint PreferredDscp() const { |
380 return rtc::DSCP_DEFAULT; | 380 return rtc::DSCP_DEFAULT; |
381 } | 381 } |
382 // Called when a RTP packet is received. | 382 // Called when a RTP packet is received. |
383 virtual void OnPacketReceived(rtc::Buffer* packet, | 383 virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet, |
384 const rtc::PacketTime& packet_time) = 0; | 384 const rtc::PacketTime& packet_time) = 0; |
385 // Called when a RTCP packet is received. | 385 // Called when a RTCP packet is received. |
386 virtual void OnRtcpReceived(rtc::Buffer* packet, | 386 virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, |
387 const rtc::PacketTime& packet_time) = 0; | 387 const rtc::PacketTime& packet_time) = 0; |
388 // Called when the socket's ability to send has changed. | 388 // Called when the socket's ability to send has changed. |
389 virtual void OnReadyToSend(bool ready) = 0; | 389 virtual void OnReadyToSend(bool ready) = 0; |
390 // Creates a new outgoing media stream with SSRCs and CNAME as described | 390 // Creates a new outgoing media stream with SSRCs and CNAME as described |
391 // by sp. | 391 // by sp. |
392 virtual bool AddSendStream(const StreamParams& sp) = 0; | 392 virtual bool AddSendStream(const StreamParams& sp) = 0; |
393 // Removes an outgoing media stream. | 393 // Removes an outgoing media stream. |
394 // ssrc must be the first SSRC of the media stream if the stream uses | 394 // ssrc must be the first SSRC of the media stream if the stream uses |
395 // multiple SSRCs. | 395 // multiple SSRCs. |
396 virtual bool RemoveSendStream(uint32_t ssrc) = 0; | 396 virtual bool RemoveSendStream(uint32_t ssrc) = 0; |
397 // Creates a new incoming media stream with SSRCs and CNAME as described | 397 // Creates a new incoming media stream with SSRCs and CNAME as described |
398 // by sp. | 398 // by sp. |
399 virtual bool AddRecvStream(const StreamParams& sp) = 0; | 399 virtual bool AddRecvStream(const StreamParams& sp) = 0; |
400 // Removes an incoming media stream. | 400 // Removes an incoming media stream. |
401 // ssrc must be the first SSRC of the media stream if the stream uses | 401 // ssrc must be the first SSRC of the media stream if the stream uses |
402 // multiple SSRCs. | 402 // multiple SSRCs. |
403 virtual bool RemoveRecvStream(uint32_t ssrc) = 0; | 403 virtual bool RemoveRecvStream(uint32_t ssrc) = 0; |
404 | 404 |
405 // Returns the absoulte sendtime extension id value from media channel. | 405 // Returns the absoulte sendtime extension id value from media channel. |
406 virtual int GetRtpSendTimeExtnId() const { | 406 virtual int GetRtpSendTimeExtnId() const { |
407 return -1; | 407 return -1; |
408 } | 408 } |
409 | 409 |
410 // Base method to send packet using NetworkInterface. | 410 // Base method to send packet using NetworkInterface. |
411 bool SendPacket(rtc::Buffer* packet, const rtc::PacketOptions& options) { | 411 bool SendPacket(rtc::CopyOnWriteBuffer* packet, |
| 412 const rtc::PacketOptions& options) { |
412 return DoSendPacket(packet, false, options); | 413 return DoSendPacket(packet, false, options); |
413 } | 414 } |
414 | 415 |
415 bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options) { | 416 bool SendRtcp(rtc::CopyOnWriteBuffer* packet, |
| 417 const rtc::PacketOptions& options) { |
416 return DoSendPacket(packet, true, options); | 418 return DoSendPacket(packet, true, options); |
417 } | 419 } |
418 | 420 |
419 int SetOption(NetworkInterface::SocketType type, | 421 int SetOption(NetworkInterface::SocketType type, |
420 rtc::Socket::Option opt, | 422 rtc::Socket::Option opt, |
421 int option) { | 423 int option) { |
422 rtc::CritScope cs(&network_interface_crit_); | 424 rtc::CritScope cs(&network_interface_crit_); |
423 if (!network_interface_) | 425 if (!network_interface_) |
424 return -1; | 426 return -1; |
425 | 427 |
426 return network_interface_->SetOption(type, opt, option); | 428 return network_interface_->SetOption(type, opt, option); |
427 } | 429 } |
428 | 430 |
429 private: | 431 private: |
430 // This method sets DSCP |value| on both RTP and RTCP channels. | 432 // This method sets DSCP |value| on both RTP and RTCP channels. |
431 int SetDscp(rtc::DiffServCodePoint value) { | 433 int SetDscp(rtc::DiffServCodePoint value) { |
432 int ret; | 434 int ret; |
433 ret = SetOption(NetworkInterface::ST_RTP, | 435 ret = SetOption(NetworkInterface::ST_RTP, |
434 rtc::Socket::OPT_DSCP, | 436 rtc::Socket::OPT_DSCP, |
435 value); | 437 value); |
436 if (ret == 0) { | 438 if (ret == 0) { |
437 ret = SetOption(NetworkInterface::ST_RTCP, | 439 ret = SetOption(NetworkInterface::ST_RTCP, |
438 rtc::Socket::OPT_DSCP, | 440 rtc::Socket::OPT_DSCP, |
439 value); | 441 value); |
440 } | 442 } |
441 return ret; | 443 return ret; |
442 } | 444 } |
443 | 445 |
444 bool DoSendPacket(rtc::Buffer* packet, | 446 bool DoSendPacket(rtc::CopyOnWriteBuffer* packet, |
445 bool rtcp, | 447 bool rtcp, |
446 const rtc::PacketOptions& options) { | 448 const rtc::PacketOptions& options) { |
447 rtc::CritScope cs(&network_interface_crit_); | 449 rtc::CritScope cs(&network_interface_crit_); |
448 if (!network_interface_) | 450 if (!network_interface_) |
449 return false; | 451 return false; |
450 | 452 |
451 return (!rtcp) ? network_interface_->SendPacket(packet, options) | 453 return (!rtcp) ? network_interface_->SendPacket(packet, options) |
452 : network_interface_->SendRtcp(packet, options); | 454 : network_interface_->SendRtcp(packet, options); |
453 } | 455 } |
454 | 456 |
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1097 virtual bool SetRecvParameters(const DataRecvParameters& params) = 0; | 1099 virtual bool SetRecvParameters(const DataRecvParameters& params) = 0; |
1098 | 1100 |
1099 // TODO(pthatcher): Implement this. | 1101 // TODO(pthatcher): Implement this. |
1100 virtual bool GetStats(DataMediaInfo* info) { return true; } | 1102 virtual bool GetStats(DataMediaInfo* info) { return true; } |
1101 | 1103 |
1102 virtual bool SetSend(bool send) = 0; | 1104 virtual bool SetSend(bool send) = 0; |
1103 virtual bool SetReceive(bool receive) = 0; | 1105 virtual bool SetReceive(bool receive) = 0; |
1104 | 1106 |
1105 virtual bool SendData( | 1107 virtual bool SendData( |
1106 const SendDataParams& params, | 1108 const SendDataParams& params, |
1107 const rtc::Buffer& payload, | 1109 const rtc::CopyOnWriteBuffer& payload, |
1108 SendDataResult* result = NULL) = 0; | 1110 SendDataResult* result = NULL) = 0; |
1109 // Signals when data is received (params, data, len) | 1111 // Signals when data is received (params, data, len) |
1110 sigslot::signal3<const ReceiveDataParams&, | 1112 sigslot::signal3<const ReceiveDataParams&, |
1111 const char*, | 1113 const char*, |
1112 size_t> SignalDataReceived; | 1114 size_t> SignalDataReceived; |
1113 // Signal when the media channel is ready to send the stream. Arguments are: | 1115 // Signal when the media channel is ready to send the stream. Arguments are: |
1114 // writable(bool) | 1116 // writable(bool) |
1115 sigslot::signal1<bool> SignalReadyToSend; | 1117 sigslot::signal1<bool> SignalReadyToSend; |
1116 // Signal for notifying that the remote side has closed the DataChannel. | 1118 // Signal for notifying that the remote side has closed the DataChannel. |
1117 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; | 1119 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; |
1118 }; | 1120 }; |
1119 | 1121 |
1120 } // namespace cricket | 1122 } // namespace cricket |
1121 | 1123 |
1122 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 1124 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
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