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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2004 Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#include "talk/session/media/channel.h"
29
30#include "talk/base/buffer.h"
31#include "talk/base/byteorder.h"
32#include "talk/base/common.h"
33#include "talk/base/logging.h"
34#include "talk/media/base/rtputils.h"
35#include "talk/p2p/base/transportchannel.h"
36#include "talk/session/media/channelmanager.h"
37#include "talk/session/media/mediamessages.h"
38#include "talk/session/media/rtcpmuxfilter.h"
39#include "talk/session/media/typingmonitor.h"
40
41
42namespace cricket {
43
44enum {
45 MSG_ENABLE = 1,
46 MSG_DISABLE,
47 MSG_MUTESTREAM,
48 MSG_ISSTREAMMUTED,
49 MSG_SETREMOTECONTENT,
50 MSG_SETLOCALCONTENT,
51 MSG_EARLYMEDIATIMEOUT,
52 MSG_CANINSERTDTMF,
53 MSG_INSERTDTMF,
54 MSG_GETSTATS,
55 MSG_SETRENDERER,
56 MSG_ADDRECVSTREAM,
57 MSG_REMOVERECVSTREAM,
58 MSG_SETRINGBACKTONE,
59 MSG_PLAYRINGBACKTONE,
60 MSG_SETMAXSENDBANDWIDTH,
61 MSG_ADDSCREENCAST,
62 MSG_REMOVESCREENCAST,
63 MSG_SENDINTRAFRAME,
64 MSG_REQUESTINTRAFRAME,
65 MSG_SCREENCASTWINDOWEVENT,
66 MSG_RTPPACKET,
67 MSG_RTCPPACKET,
68 MSG_CHANNEL_ERROR,
69 MSG_SETCHANNELOPTIONS,
70 MSG_SCALEVOLUME,
71 MSG_HANDLEVIEWREQUEST,
72 MSG_READYTOSENDDATA,
73 MSG_SENDDATA,
74 MSG_DATARECEIVED,
75 MSG_SETCAPTURER,
76 MSG_ISSCREENCASTING,
77 MSG_SCREENCASTFPS,
78 MSG_SETSCREENCASTFACTORY,
79 MSG_FIRSTPACKETRECEIVED,
80 MSG_SESSION_ERROR,
81};
82
83// Value specified in RFC 5764.
84static const char kDtlsSrtpExporterLabel[] = "EXTRACTOR-dtls_srtp";
85
86static const int kAgcMinus10db = -10;
87
88// TODO(hellner): use the device manager for creation of screen capturers when
89// the cl enabling it has landed.
90class NullScreenCapturerFactory : public VideoChannel::ScreenCapturerFactory {
91 public:
92 VideoCapturer* CreateScreenCapturer(const ScreencastId& window) {
93 return NULL;
94 }
95};
96
97
98VideoChannel::ScreenCapturerFactory* CreateScreenCapturerFactory() {
99 return new NullScreenCapturerFactory();
100}
101
102struct SetContentData : public talk_base::MessageData {
103 SetContentData(const MediaContentDescription* content, ContentAction action)
104 : content(content),
105 action(action),
106 result(false) {
107 }
108 const MediaContentDescription* content;
109 ContentAction action;
110 bool result;
111};
112
113struct SetBandwidthData : public talk_base::MessageData {
114 explicit SetBandwidthData(int value) : value(value), result(false) {}
115 int value;
116 bool result;
117};
118
119struct SetRingbackToneMessageData : public talk_base::MessageData {
120 SetRingbackToneMessageData(const void* b, int l)
121 : buf(b),
122 len(l),
123 result(false) {
124 }
125 const void* buf;
126 int len;
127 bool result;
128};
129
130struct PlayRingbackToneMessageData : public talk_base::MessageData {
131 PlayRingbackToneMessageData(uint32 s, bool p, bool l)
132 : ssrc(s),
133 play(p),
134 loop(l),
135 result(false) {
136 }
137 uint32 ssrc;
138 bool play;
139 bool loop;
140 bool result;
141};
142typedef talk_base::TypedMessageData<bool> BoolMessageData;
143struct DtmfMessageData : public talk_base::MessageData {
144 DtmfMessageData(uint32 ssrc, int event, int duration, int flags)
145 : ssrc(ssrc),
146 event(event),
147 duration(duration),
148 flags(flags),
149 result(false) {
150 }
151 uint32 ssrc;
152 int event;
153 int duration;
154 int flags;
155 bool result;
156};
157struct ScaleVolumeMessageData : public talk_base::MessageData {
158 ScaleVolumeMessageData(uint32 s, double l, double r)
159 : ssrc(s),
160 left(l),
161 right(r),
162 result(false) {
163 }
164 uint32 ssrc;
165 double left;
166 double right;
167 bool result;
168};
169
170struct VoiceStatsMessageData : public talk_base::MessageData {
171 explicit VoiceStatsMessageData(VoiceMediaInfo* stats)
172 : result(false),
173 stats(stats) {
174 }
175 bool result;
176 VoiceMediaInfo* stats;
177};
178
179struct VideoStatsMessageData : public talk_base::MessageData {
180 explicit VideoStatsMessageData(VideoMediaInfo* stats)
181 : result(false),
182 stats(stats) {
183 }
184 bool result;
185 VideoMediaInfo* stats;
186};
187
188struct PacketMessageData : public talk_base::MessageData {
189 talk_base::Buffer packet;
190};
191
192struct AudioRenderMessageData: public talk_base::MessageData {
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000193 AudioRenderMessageData(uint32 s, AudioRenderer* r, bool l)
194 : ssrc(s), renderer(r), is_local(l), result(false) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000195 uint32 ssrc;
196 AudioRenderer* renderer;
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000197 bool is_local;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000198 bool result;
199};
200
201struct VideoRenderMessageData : public talk_base::MessageData {
202 VideoRenderMessageData(uint32 s, VideoRenderer* r) : ssrc(s), renderer(r) {}
203 uint32 ssrc;
204 VideoRenderer* renderer;
205};
206
207struct AddScreencastMessageData : public talk_base::MessageData {
208 AddScreencastMessageData(uint32 s, const ScreencastId& id)
209 : ssrc(s),
210 window_id(id),
211 result(NULL) {
212 }
213 uint32 ssrc;
214 ScreencastId window_id;
215 VideoCapturer* result;
216};
217
218struct RemoveScreencastMessageData : public talk_base::MessageData {
219 explicit RemoveScreencastMessageData(uint32 s) : ssrc(s), result(false) {}
220 uint32 ssrc;
221 bool result;
222};
223
224struct ScreencastEventMessageData : public talk_base::MessageData {
225 ScreencastEventMessageData(uint32 s, talk_base::WindowEvent we)
226 : ssrc(s),
227 event(we) {
228 }
229 uint32 ssrc;
230 talk_base::WindowEvent event;
231};
232
233struct ViewRequestMessageData : public talk_base::MessageData {
234 explicit ViewRequestMessageData(const ViewRequest& r)
235 : request(r),
236 result(false) {
237 }
238 ViewRequest request;
239 bool result;
240};
241
242struct VoiceChannelErrorMessageData : public talk_base::MessageData {
243 VoiceChannelErrorMessageData(uint32 in_ssrc,
244 VoiceMediaChannel::Error in_error)
245 : ssrc(in_ssrc),
246 error(in_error) {
247 }
248 uint32 ssrc;
249 VoiceMediaChannel::Error error;
250};
251
252struct VideoChannelErrorMessageData : public talk_base::MessageData {
253 VideoChannelErrorMessageData(uint32 in_ssrc,
254 VideoMediaChannel::Error in_error)
255 : ssrc(in_ssrc),
256 error(in_error) {
257 }
258 uint32 ssrc;
259 VideoMediaChannel::Error error;
260};
261
262struct DataChannelErrorMessageData : public talk_base::MessageData {
263 DataChannelErrorMessageData(uint32 in_ssrc,
264 DataMediaChannel::Error in_error)
265 : ssrc(in_ssrc),
266 error(in_error) {}
267 uint32 ssrc;
268 DataMediaChannel::Error error;
269};
270
271struct SessionErrorMessageData : public talk_base::MessageData {
272 explicit SessionErrorMessageData(cricket::BaseSession::Error error)
273 : error_(error) {}
274
275 BaseSession::Error error_;
276};
277
278struct SsrcMessageData : public talk_base::MessageData {
279 explicit SsrcMessageData(uint32 ssrc) : ssrc(ssrc), result(false) {}
280 uint32 ssrc;
281 bool result;
282};
283
284struct StreamMessageData : public talk_base::MessageData {
285 explicit StreamMessageData(const StreamParams& in_sp)
286 : sp(in_sp),
287 result(false) {
288 }
289 StreamParams sp;
290 bool result;
291};
292
293struct MuteStreamData : public talk_base::MessageData {
294 MuteStreamData(uint32 ssrc, bool mute)
295 : ssrc(ssrc), mute(mute), result(false) {}
296 uint32 ssrc;
297 bool mute;
298 bool result;
299};
300
301struct AudioOptionsMessageData : public talk_base::MessageData {
302 explicit AudioOptionsMessageData(const AudioOptions& options)
303 : options(options),
304 result(false) {
305 }
306 AudioOptions options;
307 bool result;
308};
309
310struct VideoOptionsMessageData : public talk_base::MessageData {
311 explicit VideoOptionsMessageData(const VideoOptions& options)
312 : options(options),
313 result(false) {
314 }
315 VideoOptions options;
316 bool result;
317};
318
319struct SetCapturerMessageData : public talk_base::MessageData {
320 SetCapturerMessageData(uint32 s, VideoCapturer* c)
321 : ssrc(s),
322 capturer(c),
323 result(false) {
324 }
325 uint32 ssrc;
326 VideoCapturer* capturer;
327 bool result;
328};
329
330struct IsScreencastingMessageData : public talk_base::MessageData {
331 IsScreencastingMessageData()
332 : result(false) {
333 }
334 bool result;
335};
336
337struct ScreencastFpsMessageData : public talk_base::MessageData {
338 explicit ScreencastFpsMessageData(uint32 s)
339 : ssrc(s), result(0) {
340 }
341 uint32 ssrc;
342 int result;
343};
344
345struct SetScreenCaptureFactoryMessageData : public talk_base::MessageData {
346 explicit SetScreenCaptureFactoryMessageData(
347 VideoChannel::ScreenCapturerFactory* f)
348 : screencapture_factory(f) {
349 }
350 VideoChannel::ScreenCapturerFactory* screencapture_factory;
351};
352
353static const char* PacketType(bool rtcp) {
354 return (!rtcp) ? "RTP" : "RTCP";
355}
356
357static bool ValidPacket(bool rtcp, const talk_base::Buffer* packet) {
358 // Check the packet size. We could check the header too if needed.
359 return (packet &&
360 packet->length() >= (!rtcp ? kMinRtpPacketLen : kMinRtcpPacketLen) &&
361 packet->length() <= kMaxRtpPacketLen);
362}
363
364static bool IsReceiveContentDirection(MediaContentDirection direction) {
365 return direction == MD_SENDRECV || direction == MD_RECVONLY;
366}
367
368static bool IsSendContentDirection(MediaContentDirection direction) {
369 return direction == MD_SENDRECV || direction == MD_SENDONLY;
370}
371
372static const MediaContentDescription* GetContentDescription(
373 const ContentInfo* cinfo) {
374 if (cinfo == NULL)
375 return NULL;
376 return static_cast<const MediaContentDescription*>(cinfo->description);
377}
378
379BaseChannel::BaseChannel(talk_base::Thread* thread,
380 MediaEngineInterface* media_engine,
381 MediaChannel* media_channel, BaseSession* session,
382 const std::string& content_name, bool rtcp)
383 : worker_thread_(thread),
384 media_engine_(media_engine),
385 session_(session),
386 media_channel_(media_channel),
387 content_name_(content_name),
388 rtcp_(rtcp),
389 transport_channel_(NULL),
390 rtcp_transport_channel_(NULL),
391 enabled_(false),
392 writable_(false),
393 rtp_ready_to_send_(false),
394 rtcp_ready_to_send_(false),
395 optimistic_data_send_(false),
396 was_ever_writable_(false),
397 local_content_direction_(MD_INACTIVE),
398 remote_content_direction_(MD_INACTIVE),
399 has_received_packet_(false),
400 dtls_keyed_(false),
401 secure_required_(false) {
402 ASSERT(worker_thread_ == talk_base::Thread::Current());
403 LOG(LS_INFO) << "Created channel for " << content_name;
404}
405
406BaseChannel::~BaseChannel() {
407 ASSERT(worker_thread_ == talk_base::Thread::Current());
408 StopConnectionMonitor();
409 FlushRtcpMessages(); // Send any outstanding RTCP packets.
410 Clear(); // eats any outstanding messages or packets
411 // We must destroy the media channel before the transport channel, otherwise
412 // the media channel may try to send on the dead transport channel. NULLing
413 // is not an effective strategy since the sends will come on another thread.
414 delete media_channel_;
415 set_rtcp_transport_channel(NULL);
416 if (transport_channel_ != NULL)
417 session_->DestroyChannel(content_name_, transport_channel_->component());
418 LOG(LS_INFO) << "Destroyed channel";
419}
420
421bool BaseChannel::Init(TransportChannel* transport_channel,
422 TransportChannel* rtcp_transport_channel) {
423 if (transport_channel == NULL) {
424 return false;
425 }
426 if (rtcp() && rtcp_transport_channel == NULL) {
427 return false;
428 }
429 transport_channel_ = transport_channel;
430
431 if (!SetDtlsSrtpCiphers(transport_channel_, false)) {
432 return false;
433 }
434
435 media_channel_->SetInterface(this);
436 transport_channel_->SignalWritableState.connect(
437 this, &BaseChannel::OnWritableState);
438 transport_channel_->SignalReadPacket.connect(
439 this, &BaseChannel::OnChannelRead);
440 transport_channel_->SignalReadyToSend.connect(
441 this, &BaseChannel::OnReadyToSend);
442
443 session_->SignalNewLocalDescription.connect(
444 this, &BaseChannel::OnNewLocalDescription);
445 session_->SignalNewRemoteDescription.connect(
446 this, &BaseChannel::OnNewRemoteDescription);
447
448 set_rtcp_transport_channel(rtcp_transport_channel);
449 return true;
450}
451
452// Can be called from thread other than worker thread
453bool BaseChannel::Enable(bool enable) {
454 Send(enable ? MSG_ENABLE : MSG_DISABLE);
455 return true;
456}
457
458// Can be called from thread other than worker thread
459bool BaseChannel::MuteStream(uint32 ssrc, bool mute) {
460 MuteStreamData data(ssrc, mute);
461 Send(MSG_MUTESTREAM, &data);
462 return data.result;
463}
464
465bool BaseChannel::IsStreamMuted(uint32 ssrc) {
466 SsrcMessageData data(ssrc);
467 Send(MSG_ISSTREAMMUTED, &data);
468 return data.result;
469}
470
471bool BaseChannel::AddRecvStream(const StreamParams& sp) {
472 StreamMessageData data(sp);
473 Send(MSG_ADDRECVSTREAM, &data);
474 return data.result;
475}
476
477bool BaseChannel::RemoveRecvStream(uint32 ssrc) {
478 SsrcMessageData data(ssrc);
479 Send(MSG_REMOVERECVSTREAM, &data);
480 return data.result;
481}
482
483bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
484 ContentAction action) {
485 SetContentData data(content, action);
486 Send(MSG_SETLOCALCONTENT, &data);
487 return data.result;
488}
489
490bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
491 ContentAction action) {
492 SetContentData data(content, action);
493 Send(MSG_SETREMOTECONTENT, &data);
494 return data.result;
495}
496
497bool BaseChannel::SetMaxSendBandwidth(int max_bandwidth) {
498 SetBandwidthData data(max_bandwidth);
499 Send(MSG_SETMAXSENDBANDWIDTH, &data);
500 return data.result;
501}
502
503void BaseChannel::StartConnectionMonitor(int cms) {
504 socket_monitor_.reset(new SocketMonitor(transport_channel_,
505 worker_thread(),
506 talk_base::Thread::Current()));
507 socket_monitor_->SignalUpdate.connect(
508 this, &BaseChannel::OnConnectionMonitorUpdate);
509 socket_monitor_->Start(cms);
510}
511
512void BaseChannel::StopConnectionMonitor() {
513 if (socket_monitor_) {
514 socket_monitor_->Stop();
515 socket_monitor_.reset();
516 }
517}
518
519void BaseChannel::set_rtcp_transport_channel(TransportChannel* channel) {
520 if (rtcp_transport_channel_ != channel) {
521 if (rtcp_transport_channel_) {
522 session_->DestroyChannel(
523 content_name_, rtcp_transport_channel_->component());
524 }
525 rtcp_transport_channel_ = channel;
526 if (rtcp_transport_channel_) {
527 // TODO(juberti): Propagate this error code
528 VERIFY(SetDtlsSrtpCiphers(rtcp_transport_channel_, true));
529 rtcp_transport_channel_->SignalWritableState.connect(
530 this, &BaseChannel::OnWritableState);
531 rtcp_transport_channel_->SignalReadPacket.connect(
532 this, &BaseChannel::OnChannelRead);
533 rtcp_transport_channel_->SignalReadyToSend.connect(
534 this, &BaseChannel::OnReadyToSend);
535 }
536 }
537}
538
539bool BaseChannel::IsReadyToReceive() const {
540 // Receive data if we are enabled and have local content,
541 return enabled() && IsReceiveContentDirection(local_content_direction_);
542}
543
544bool BaseChannel::IsReadyToSend() const {
545 // Send outgoing data if we are enabled, have local and remote content,
546 // and we have had some form of connectivity.
547 return enabled() &&
548 IsReceiveContentDirection(remote_content_direction_) &&
549 IsSendContentDirection(local_content_direction_) &&
550 was_ever_writable();
551}
552
553bool BaseChannel::SendPacket(talk_base::Buffer* packet) {
554 return SendPacket(false, packet);
555}
556
557bool BaseChannel::SendRtcp(talk_base::Buffer* packet) {
558 return SendPacket(true, packet);
559}
560
561int BaseChannel::SetOption(SocketType type, talk_base::Socket::Option opt,
562 int value) {
563 switch (type) {
564 case ST_RTP: return transport_channel_->SetOption(opt, value);
565 case ST_RTCP: return rtcp_transport_channel_->SetOption(opt, value);
566 default: return -1;
567 }
568}
569
570void BaseChannel::OnWritableState(TransportChannel* channel) {
571 ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
572 if (transport_channel_->writable()
573 && (!rtcp_transport_channel_ || rtcp_transport_channel_->writable())) {
574 ChannelWritable_w();
575 } else {
576 ChannelNotWritable_w();
577 }
578}
579
580void BaseChannel::OnChannelRead(TransportChannel* channel,
581 const char* data, size_t len, int flags) {
582 // OnChannelRead gets called from P2PSocket; now pass data to MediaEngine
583 ASSERT(worker_thread_ == talk_base::Thread::Current());
584
585 // When using RTCP multiplexing we might get RTCP packets on the RTP
586 // transport. We feed RTP traffic into the demuxer to determine if it is RTCP.
587 bool rtcp = PacketIsRtcp(channel, data, len);
588 talk_base::Buffer packet(data, len);
589 HandlePacket(rtcp, &packet);
590}
591
592void BaseChannel::OnReadyToSend(TransportChannel* channel) {
593 SetReadyToSend(channel, true);
594}
595
596void BaseChannel::SetReadyToSend(TransportChannel* channel, bool ready) {
597 ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
598 if (channel == transport_channel_) {
599 rtp_ready_to_send_ = ready;
600 }
601 if (channel == rtcp_transport_channel_) {
602 rtcp_ready_to_send_ = ready;
603 }
604
605 if (!ready) {
606 // Notify the MediaChannel when either rtp or rtcp channel can't send.
607 media_channel_->OnReadyToSend(false);
608 } else if (rtp_ready_to_send_ &&
609 // In the case of rtcp mux |rtcp_transport_channel_| will be null.
610 (rtcp_ready_to_send_ || !rtcp_transport_channel_)) {
611 // Notify the MediaChannel when both rtp and rtcp channel can send.
612 media_channel_->OnReadyToSend(true);
613 }
614}
615
616bool BaseChannel::PacketIsRtcp(const TransportChannel* channel,
617 const char* data, size_t len) {
618 return (channel == rtcp_transport_channel_ ||
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000619 rtcp_mux_filter_.DemuxRtcp(data, static_cast<int>(len)));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000620}
621
622bool BaseChannel::SendPacket(bool rtcp, talk_base::Buffer* packet) {
623 // Unless we're sending optimistically, we only allow packets through when we
624 // are completely writable.
625 if (!optimistic_data_send_ && !writable_) {
626 return false;
627 }
628
629 // SendPacket gets called from MediaEngine, typically on an encoder thread.
630 // If the thread is not our worker thread, we will post to our worker
631 // so that the real work happens on our worker. This avoids us having to
632 // synchronize access to all the pieces of the send path, including
633 // SRTP and the inner workings of the transport channels.
634 // The only downside is that we can't return a proper failure code if
635 // needed. Since UDP is unreliable anyway, this should be a non-issue.
636 if (talk_base::Thread::Current() != worker_thread_) {
637 // Avoid a copy by transferring the ownership of the packet data.
638 int message_id = (!rtcp) ? MSG_RTPPACKET : MSG_RTCPPACKET;
639 PacketMessageData* data = new PacketMessageData;
640 packet->TransferTo(&data->packet);
641 worker_thread_->Post(this, message_id, data);
642 return true;
643 }
644
645 // Now that we are on the correct thread, ensure we have a place to send this
646 // packet before doing anything. (We might get RTCP packets that we don't
647 // intend to send.) If we've negotiated RTCP mux, send RTCP over the RTP
648 // transport.
649 TransportChannel* channel = (!rtcp || rtcp_mux_filter_.IsActive()) ?
650 transport_channel_ : rtcp_transport_channel_;
651 if (!channel || (!optimistic_data_send_ && !channel->writable())) {
652 return false;
653 }
654
655 // Protect ourselves against crazy data.
656 if (!ValidPacket(rtcp, packet)) {
657 LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " "
658 << PacketType(rtcp) << " packet: wrong size="
659 << packet->length();
660 return false;
661 }
662
663 // Signal to the media sink before protecting the packet.
664 {
665 talk_base::CritScope cs(&signal_send_packet_cs_);
666 SignalSendPacketPreCrypto(packet->data(), packet->length(), rtcp);
667 }
668
669 // Protect if needed.
670 if (srtp_filter_.IsActive()) {
671 bool res;
672 char* data = packet->data();
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000673 int len = static_cast<int>(packet->length());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000674 if (!rtcp) {
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000675 res = srtp_filter_.ProtectRtp(data, len,
676 static_cast<int>(packet->capacity()), &len);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000677 if (!res) {
678 int seq_num = -1;
679 uint32 ssrc = 0;
680 GetRtpSeqNum(data, len, &seq_num);
681 GetRtpSsrc(data, len, &ssrc);
682 LOG(LS_ERROR) << "Failed to protect " << content_name_
683 << " RTP packet: size=" << len
684 << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
685 return false;
686 }
687 } else {
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000688 res = srtp_filter_.ProtectRtcp(data, len,
689 static_cast<int>(packet->capacity()),
690 &len);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000691 if (!res) {
692 int type = -1;
693 GetRtcpType(data, len, &type);
694 LOG(LS_ERROR) << "Failed to protect " << content_name_
695 << " RTCP packet: size=" << len << ", type=" << type;
696 return false;
697 }
698 }
699
700 // Update the length of the packet now that we've added the auth tag.
701 packet->SetLength(len);
702 } else if (secure_required_) {
703 // This is a double check for something that supposedly can't happen.
704 LOG(LS_ERROR) << "Can't send outgoing " << PacketType(rtcp)
705 << " packet when SRTP is inactive and crypto is required";
706
707 ASSERT(false);
708 return false;
709 }
710
711 // Signal to the media sink after protecting the packet.
712 {
713 talk_base::CritScope cs(&signal_send_packet_cs_);
714 SignalSendPacketPostCrypto(packet->data(), packet->length(), rtcp);
715 }
716
717 // Bon voyage.
718 int ret = channel->SendPacket(packet->data(), packet->length(),
719 (secure() && secure_dtls()) ? PF_SRTP_BYPASS : 0);
720 if (ret != static_cast<int>(packet->length())) {
721 if (channel->GetError() == EWOULDBLOCK) {
722 LOG(LS_WARNING) << "Got EWOULDBLOCK from socket.";
723 SetReadyToSend(channel, false);
724 }
725 return false;
726 }
727 return true;
728}
729
730bool BaseChannel::WantsPacket(bool rtcp, talk_base::Buffer* packet) {
731 // Protect ourselves against crazy data.
732 if (!ValidPacket(rtcp, packet)) {
733 LOG(LS_ERROR) << "Dropping incoming " << content_name_ << " "
734 << PacketType(rtcp) << " packet: wrong size="
735 << packet->length();
736 return false;
737 }
738 // If this channel is suppose to handle RTP data, that is determined by
739 // checking against ssrc filter. This is necessary to do it here to avoid
740 // double decryption.
741 if (ssrc_filter_.IsActive() &&
742 !ssrc_filter_.DemuxPacket(packet->data(), packet->length(), rtcp)) {
743 return false;
744 }
745
746 return true;
747}
748
749void BaseChannel::HandlePacket(bool rtcp, talk_base::Buffer* packet) {
750 if (!WantsPacket(rtcp, packet)) {
751 return;
752 }
753
754 if (!has_received_packet_) {
755 has_received_packet_ = true;
756 signaling_thread()->Post(this, MSG_FIRSTPACKETRECEIVED);
757 }
758
759 // Signal to the media sink before unprotecting the packet.
760 {
761 talk_base::CritScope cs(&signal_recv_packet_cs_);
762 SignalRecvPacketPostCrypto(packet->data(), packet->length(), rtcp);
763 }
764
765 // Unprotect the packet, if needed.
766 if (srtp_filter_.IsActive()) {
767 char* data = packet->data();
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000768 int len = static_cast<int>(packet->length());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000769 bool res;
770 if (!rtcp) {
771 res = srtp_filter_.UnprotectRtp(data, len, &len);
772 if (!res) {
773 int seq_num = -1;
774 uint32 ssrc = 0;
775 GetRtpSeqNum(data, len, &seq_num);
776 GetRtpSsrc(data, len, &ssrc);
777 LOG(LS_ERROR) << "Failed to unprotect " << content_name_
778 << " RTP packet: size=" << len
779 << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
780 return;
781 }
782 } else {
783 res = srtp_filter_.UnprotectRtcp(data, len, &len);
784 if (!res) {
785 int type = -1;
786 GetRtcpType(data, len, &type);
787 LOG(LS_ERROR) << "Failed to unprotect " << content_name_
788 << " RTCP packet: size=" << len << ", type=" << type;
789 return;
790 }
791 }
792
793 packet->SetLength(len);
794 } else if (secure_required_) {
795 // Our session description indicates that SRTP is required, but we got a
796 // packet before our SRTP filter is active. This means either that
797 // a) we got SRTP packets before we received the SDES keys, in which case
798 // we can't decrypt it anyway, or
799 // b) we got SRTP packets before DTLS completed on both the RTP and RTCP
800 // channels, so we haven't yet extracted keys, even if DTLS did complete
801 // on the channel that the packets are being sent on. It's really good
802 // practice to wait for both RTP and RTCP to be good to go before sending
803 // media, to prevent weird failure modes, so it's fine for us to just eat
804 // packets here. This is all sidestepped if RTCP mux is used anyway.
805 LOG(LS_WARNING) << "Can't process incoming " << PacketType(rtcp)
806 << " packet when SRTP is inactive and crypto is required";
807 return;
808 }
809
810 // Signal to the media sink after unprotecting the packet.
811 {
812 talk_base::CritScope cs(&signal_recv_packet_cs_);
813 SignalRecvPacketPreCrypto(packet->data(), packet->length(), rtcp);
814 }
815
816 // Push it down to the media channel.
817 if (!rtcp) {
818 media_channel_->OnPacketReceived(packet);
819 } else {
820 media_channel_->OnRtcpReceived(packet);
821 }
822}
823
824void BaseChannel::OnNewLocalDescription(
825 BaseSession* session, ContentAction action) {
826 const ContentInfo* content_info =
827 GetFirstContent(session->local_description());
828 const MediaContentDescription* content_desc =
829 GetContentDescription(content_info);
830 if (content_desc && content_info && !content_info->rejected &&
831 !SetLocalContent(content_desc, action)) {
832 LOG(LS_ERROR) << "Failure in SetLocalContent with action " << action;
833 session->SetError(BaseSession::ERROR_CONTENT);
834 }
835}
836
837void BaseChannel::OnNewRemoteDescription(
838 BaseSession* session, ContentAction action) {
839 const ContentInfo* content_info =
840 GetFirstContent(session->remote_description());
841 const MediaContentDescription* content_desc =
842 GetContentDescription(content_info);
843 if (content_desc && content_info && !content_info->rejected &&
844 !SetRemoteContent(content_desc, action)) {
845 LOG(LS_ERROR) << "Failure in SetRemoteContent with action " << action;
846 session->SetError(BaseSession::ERROR_CONTENT);
847 }
848}
849
850void BaseChannel::EnableMedia_w() {
851 ASSERT(worker_thread_ == talk_base::Thread::Current());
852 if (enabled_)
853 return;
854
855 LOG(LS_INFO) << "Channel enabled";
856 enabled_ = true;
857 ChangeState();
858}
859
860void BaseChannel::DisableMedia_w() {
861 ASSERT(worker_thread_ == talk_base::Thread::Current());
862 if (!enabled_)
863 return;
864
865 LOG(LS_INFO) << "Channel disabled";
866 enabled_ = false;
867 ChangeState();
868}
869
870bool BaseChannel::MuteStream_w(uint32 ssrc, bool mute) {
871 ASSERT(worker_thread_ == talk_base::Thread::Current());
872 bool ret = media_channel()->MuteStream(ssrc, mute);
873 if (ret) {
874 if (mute)
875 muted_streams_.insert(ssrc);
876 else
877 muted_streams_.erase(ssrc);
878 }
879 return ret;
880}
881
882bool BaseChannel::IsStreamMuted_w(uint32 ssrc) {
883 ASSERT(worker_thread_ == talk_base::Thread::Current());
884 return muted_streams_.find(ssrc) != muted_streams_.end();
885}
886
887void BaseChannel::ChannelWritable_w() {
888 ASSERT(worker_thread_ == talk_base::Thread::Current());
889 if (writable_)
890 return;
891
892 LOG(LS_INFO) << "Channel socket writable ("
893 << transport_channel_->content_name() << ", "
894 << transport_channel_->component() << ")"
895 << (was_ever_writable_ ? "" : " for the first time");
896
897 std::vector<ConnectionInfo> infos;
898 transport_channel_->GetStats(&infos);
899 for (std::vector<ConnectionInfo>::const_iterator it = infos.begin();
900 it != infos.end(); ++it) {
901 if (it->best_connection) {
902 LOG(LS_INFO) << "Using " << it->local_candidate.ToSensitiveString()
903 << "->" << it->remote_candidate.ToSensitiveString();
904 break;
905 }
906 }
907
908 // If we're doing DTLS-SRTP, now is the time.
909 if (!was_ever_writable_ && ShouldSetupDtlsSrtp()) {
910 if (!SetupDtlsSrtp(false)) {
911 LOG(LS_ERROR) << "Couldn't finish DTLS-SRTP on RTP channel";
912 SessionErrorMessageData data(BaseSession::ERROR_TRANSPORT);
913 // Sent synchronously.
914 signaling_thread()->Send(this, MSG_SESSION_ERROR, &data);
915 return;
916 }
917
918 if (rtcp_transport_channel_) {
919 if (!SetupDtlsSrtp(true)) {
920 LOG(LS_ERROR) << "Couldn't finish DTLS-SRTP on RTCP channel";
921 SessionErrorMessageData data(BaseSession::ERROR_TRANSPORT);
922 // Sent synchronously.
923 signaling_thread()->Send(this, MSG_SESSION_ERROR, &data);
924 return;
925 }
926 }
927 }
928
929 was_ever_writable_ = true;
930 writable_ = true;
931 ChangeState();
932}
933
934bool BaseChannel::SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp) {
935 std::vector<std::string> ciphers;
936 // We always use the default SRTP ciphers for RTCP, but we may use different
937 // ciphers for RTP depending on the media type.
938 if (!rtcp) {
939 GetSrtpCiphers(&ciphers);
940 } else {
941 GetSupportedDefaultCryptoSuites(&ciphers);
942 }
943 return tc->SetSrtpCiphers(ciphers);
944}
945
946bool BaseChannel::ShouldSetupDtlsSrtp() const {
947 return true;
948}
949
950// This function returns true if either DTLS-SRTP is not in use
951// *or* DTLS-SRTP is successfully set up.
952bool BaseChannel::SetupDtlsSrtp(bool rtcp_channel) {
953 bool ret = false;
954
955 TransportChannel *channel = rtcp_channel ?
956 rtcp_transport_channel_ : transport_channel_;
957
958 // No DTLS
959 if (!channel->IsDtlsActive())
960 return true;
961
962 std::string selected_cipher;
963
964 if (!channel->GetSrtpCipher(&selected_cipher)) {
965 LOG(LS_ERROR) << "No DTLS-SRTP selected cipher";
966 return false;
967 }
968
969 LOG(LS_INFO) << "Installing keys from DTLS-SRTP on "
970 << content_name() << " "
971 << PacketType(rtcp_channel);
972
973 // OK, we're now doing DTLS (RFC 5764)
974 std::vector<unsigned char> dtls_buffer(SRTP_MASTER_KEY_KEY_LEN * 2 +
975 SRTP_MASTER_KEY_SALT_LEN * 2);
976
977 // RFC 5705 exporter using the RFC 5764 parameters
978 if (!channel->ExportKeyingMaterial(
979 kDtlsSrtpExporterLabel,
980 NULL, 0, false,
981 &dtls_buffer[0], dtls_buffer.size())) {
982 LOG(LS_WARNING) << "DTLS-SRTP key export failed";
983 ASSERT(false); // This should never happen
984 return false;
985 }
986
987 // Sync up the keys with the DTLS-SRTP interface
988 std::vector<unsigned char> client_write_key(SRTP_MASTER_KEY_KEY_LEN +
989 SRTP_MASTER_KEY_SALT_LEN);
990 std::vector<unsigned char> server_write_key(SRTP_MASTER_KEY_KEY_LEN +
991 SRTP_MASTER_KEY_SALT_LEN);
992 size_t offset = 0;
993 memcpy(&client_write_key[0], &dtls_buffer[offset],
994 SRTP_MASTER_KEY_KEY_LEN);
995 offset += SRTP_MASTER_KEY_KEY_LEN;
996 memcpy(&server_write_key[0], &dtls_buffer[offset],
997 SRTP_MASTER_KEY_KEY_LEN);
998 offset += SRTP_MASTER_KEY_KEY_LEN;
999 memcpy(&client_write_key[SRTP_MASTER_KEY_KEY_LEN],
1000 &dtls_buffer[offset], SRTP_MASTER_KEY_SALT_LEN);
1001 offset += SRTP_MASTER_KEY_SALT_LEN;
1002 memcpy(&server_write_key[SRTP_MASTER_KEY_KEY_LEN],
1003 &dtls_buffer[offset], SRTP_MASTER_KEY_SALT_LEN);
1004
1005 std::vector<unsigned char> *send_key, *recv_key;
1006
1007 if (channel->GetRole() == ROLE_CONTROLLING) {
1008 send_key = &server_write_key;
1009 recv_key = &client_write_key;
1010 } else {
1011 send_key = &client_write_key;
1012 recv_key = &server_write_key;
1013 }
1014
1015 if (rtcp_channel) {
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00001016 ret = srtp_filter_.SetRtcpParams(
1017 selected_cipher,
1018 &(*send_key)[0],
1019 static_cast<int>(send_key->size()),
1020 selected_cipher,
1021 &(*recv_key)[0],
1022 static_cast<int>(recv_key->size()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001023 } else {
henrike@webrtc.org28654cb2013-07-22 21:07:49 +00001024 ret = srtp_filter_.SetRtpParams(
1025 selected_cipher,
1026 &(*send_key)[0],
1027 static_cast<int>(send_key->size()),
1028 selected_cipher,
1029 &(*recv_key)[0],
1030 static_cast<int>(recv_key->size()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001031 }
1032
1033 if (!ret)
1034 LOG(LS_WARNING) << "DTLS-SRTP key installation failed";
1035 else
1036 dtls_keyed_ = true;
1037
1038 return ret;
1039}
1040
1041void BaseChannel::ChannelNotWritable_w() {
1042 ASSERT(worker_thread_ == talk_base::Thread::Current());
1043 if (!writable_)
1044 return;
1045
1046 LOG(LS_INFO) << "Channel socket not writable ("
1047 << transport_channel_->content_name() << ", "
1048 << transport_channel_->component() << ")";
1049 writable_ = false;
1050 ChangeState();
1051}
1052
1053// Sets the maximum video bandwidth for automatic bandwidth adjustment.
1054bool BaseChannel::SetMaxSendBandwidth_w(int max_bandwidth) {
1055 return media_channel()->SetSendBandwidth(true, max_bandwidth);
1056}
1057
1058bool BaseChannel::SetSrtp_w(const std::vector<CryptoParams>& cryptos,
1059 ContentAction action, ContentSource src) {
1060 bool ret = false;
1061 switch (action) {
1062 case CA_OFFER:
1063 ret = srtp_filter_.SetOffer(cryptos, src);
1064 break;
1065 case CA_PRANSWER:
1066 // If we're doing DTLS-SRTP, we don't want to update the filter
1067 // with an answer, because we already have SRTP parameters.
1068 if (transport_channel_->IsDtlsActive()) {
1069 LOG(LS_INFO) <<
1070 "Ignoring SDES answer parameters because we are using DTLS-SRTP";
1071 ret = true;
1072 } else {
1073 ret = srtp_filter_.SetProvisionalAnswer(cryptos, src);
1074 }
1075 break;
1076 case CA_ANSWER:
1077 // If we're doing DTLS-SRTP, we don't want to update the filter
1078 // with an answer, because we already have SRTP parameters.
1079 if (transport_channel_->IsDtlsActive()) {
1080 LOG(LS_INFO) <<
1081 "Ignoring SDES answer parameters because we are using DTLS-SRTP";
1082 ret = true;
1083 } else {
1084 ret = srtp_filter_.SetAnswer(cryptos, src);
1085 }
1086 break;
1087 case CA_UPDATE:
1088 // no crypto params.
1089 ret = true;
1090 break;
1091 default:
1092 break;
1093 }
1094 return ret;
1095}
1096
1097bool BaseChannel::SetRtcpMux_w(bool enable, ContentAction action,
1098 ContentSource src) {
1099 bool ret = false;
1100 switch (action) {
1101 case CA_OFFER:
1102 ret = rtcp_mux_filter_.SetOffer(enable, src);
1103 break;
1104 case CA_PRANSWER:
1105 ret = rtcp_mux_filter_.SetProvisionalAnswer(enable, src);
1106 break;
1107 case CA_ANSWER:
1108 ret = rtcp_mux_filter_.SetAnswer(enable, src);
1109 if (ret && rtcp_mux_filter_.IsActive()) {
1110 // We activated RTCP mux, close down the RTCP transport.
1111 set_rtcp_transport_channel(NULL);
1112 }
1113 break;
1114 case CA_UPDATE:
1115 // No RTCP mux info.
1116 ret = true;
1117 default:
1118 break;
1119 }
1120 // |rtcp_mux_filter_| can be active if |action| is CA_PRANSWER or
1121 // CA_ANSWER, but we only want to tear down the RTCP transport channel if we
1122 // received a final answer.
1123 if (ret && rtcp_mux_filter_.IsActive()) {
1124 // If the RTP transport is already writable, then so are we.
1125 if (transport_channel_->writable()) {
1126 ChannelWritable_w();
1127 }
1128 }
1129
1130 return ret;
1131}
1132
1133bool BaseChannel::AddRecvStream_w(const StreamParams& sp) {
1134 ASSERT(worker_thread() == talk_base::Thread::Current());
1135 if (!media_channel()->AddRecvStream(sp))
1136 return false;
1137
1138 return ssrc_filter_.AddStream(sp);
1139}
1140
1141bool BaseChannel::RemoveRecvStream_w(uint32 ssrc) {
1142 ASSERT(worker_thread() == talk_base::Thread::Current());
1143 ssrc_filter_.RemoveStream(ssrc);
1144 return media_channel()->RemoveRecvStream(ssrc);
1145}
1146
1147bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
1148 ContentAction action) {
1149 if (!VERIFY(action == CA_OFFER || action == CA_ANSWER ||
1150 action == CA_PRANSWER || action == CA_UPDATE))
1151 return false;
1152
1153 // If this is an update, streams only contain streams that have changed.
1154 if (action == CA_UPDATE) {
1155 for (StreamParamsVec::const_iterator it = streams.begin();
1156 it != streams.end(); ++it) {
1157 StreamParams existing_stream;
1158 bool stream_exist = GetStreamByIds(local_streams_, it->groupid,
1159 it->id, &existing_stream);
1160 if (!stream_exist && it->has_ssrcs()) {
1161 if (media_channel()->AddSendStream(*it)) {
1162 local_streams_.push_back(*it);
1163 LOG(LS_INFO) << "Add send stream ssrc: " << it->first_ssrc();
1164 } else {
1165 LOG(LS_INFO) << "Failed to add send stream ssrc: "
1166 << it->first_ssrc();
1167 return false;
1168 }
1169 } else if (stream_exist && !it->has_ssrcs()) {
1170 if (!media_channel()->RemoveSendStream(existing_stream.first_ssrc())) {
1171 LOG(LS_ERROR) << "Failed to remove send stream with ssrc "
1172 << it->first_ssrc() << ".";
1173 return false;
1174 }
1175 RemoveStreamBySsrc(&local_streams_, existing_stream.first_ssrc());
1176 } else {
1177 LOG(LS_WARNING) << "Ignore unsupported stream update";
1178 }
1179 }
1180 return true;
1181 }
1182 // Else streams are all the streams we want to send.
1183
1184 // Check for streams that have been removed.
1185 bool ret = true;
1186 for (StreamParamsVec::const_iterator it = local_streams_.begin();
1187 it != local_streams_.end(); ++it) {
1188 if (!GetStreamBySsrc(streams, it->first_ssrc(), NULL)) {
1189 if (!media_channel()->RemoveSendStream(it->first_ssrc())) {
1190 LOG(LS_ERROR) << "Failed to remove send stream with ssrc "
1191 << it->first_ssrc() << ".";
1192 ret = false;
1193 }
1194 }
1195 }
1196 // Check for new streams.
1197 for (StreamParamsVec::const_iterator it = streams.begin();
1198 it != streams.end(); ++it) {
1199 if (!GetStreamBySsrc(local_streams_, it->first_ssrc(), NULL)) {
1200 if (media_channel()->AddSendStream(*it)) {
1201 LOG(LS_INFO) << "Add send ssrc: " << it->ssrcs[0];
1202 } else {
1203 LOG(LS_INFO) << "Failed to add send stream ssrc: " << it->first_ssrc();
1204 ret = false;
1205 }
1206 }
1207 }
1208 local_streams_ = streams;
1209 return ret;
1210}
1211
1212bool BaseChannel::UpdateRemoteStreams_w(
1213 const std::vector<StreamParams>& streams,
1214 ContentAction action) {
1215 if (!VERIFY(action == CA_OFFER || action == CA_ANSWER ||
1216 action == CA_PRANSWER || action == CA_UPDATE))
1217 return false;
1218
1219 // If this is an update, streams only contain streams that have changed.
1220 if (action == CA_UPDATE) {
1221 for (StreamParamsVec::const_iterator it = streams.begin();
1222 it != streams.end(); ++it) {
1223 StreamParams existing_stream;
1224 bool stream_exists = GetStreamByIds(remote_streams_, it->groupid,
1225 it->id, &existing_stream);
1226 if (!stream_exists && it->has_ssrcs()) {
1227 if (AddRecvStream_w(*it)) {
1228 remote_streams_.push_back(*it);
1229 LOG(LS_INFO) << "Add remote stream ssrc: " << it->first_ssrc();
1230 } else {
1231 LOG(LS_INFO) << "Failed to add remote stream ssrc: "
1232 << it->first_ssrc();
1233 return false;
1234 }
1235 } else if (stream_exists && !it->has_ssrcs()) {
1236 if (!RemoveRecvStream_w(existing_stream.first_ssrc())) {
1237 LOG(LS_ERROR) << "Failed to remove remote stream with ssrc "
1238 << it->first_ssrc() << ".";
1239 return false;
1240 }
1241 RemoveStreamBySsrc(&remote_streams_, existing_stream.first_ssrc());
1242 } else {
1243 LOG(LS_WARNING) << "Ignore unsupported stream update."
1244 << " Stream exists? " << stream_exists
1245 << " existing stream = " << existing_stream.ToString()
1246 << " new stream = " << it->ToString();
1247 }
1248 }
1249 return true;
1250 }
1251 // Else streams are all the streams we want to receive.
1252
1253 // Check for streams that have been removed.
1254 bool ret = true;
1255 for (StreamParamsVec::const_iterator it = remote_streams_.begin();
1256 it != remote_streams_.end(); ++it) {
1257 if (!GetStreamBySsrc(streams, it->first_ssrc(), NULL)) {
1258 if (!RemoveRecvStream_w(it->first_ssrc())) {
1259 LOG(LS_ERROR) << "Failed to remove remote stream with ssrc "
1260 << it->first_ssrc() << ".";
1261 ret = false;
1262 }
1263 }
1264 }
1265 // Check for new streams.
1266 for (StreamParamsVec::const_iterator it = streams.begin();
1267 it != streams.end(); ++it) {
1268 if (!GetStreamBySsrc(remote_streams_, it->first_ssrc(), NULL)) {
1269 if (AddRecvStream_w(*it)) {
1270 LOG(LS_INFO) << "Add remote ssrc: " << it->ssrcs[0];
1271 } else {
1272 LOG(LS_INFO) << "Failed to add remote stream ssrc: "
1273 << it->first_ssrc();
1274 ret = false;
1275 }
1276 }
1277 }
1278 remote_streams_ = streams;
1279 return ret;
1280}
1281
1282bool BaseChannel::SetBaseLocalContent_w(const MediaContentDescription* content,
1283 ContentAction action) {
1284 // Cache secure_required_ for belt and suspenders check on SendPacket
1285 secure_required_ = content->crypto_required();
1286 bool ret = UpdateLocalStreams_w(content->streams(), action);
1287 // Set local SRTP parameters (what we will encrypt with).
1288 ret &= SetSrtp_w(content->cryptos(), action, CS_LOCAL);
1289 // Set local RTCP mux parameters.
1290 ret &= SetRtcpMux_w(content->rtcp_mux(), action, CS_LOCAL);
1291 // Set local RTP header extensions.
1292 if (content->rtp_header_extensions_set()) {
1293 ret &= media_channel()->SetRecvRtpHeaderExtensions(
1294 content->rtp_header_extensions());
1295 }
1296 set_local_content_direction(content->direction());
1297 return ret;
1298}
1299
1300bool BaseChannel::SetBaseRemoteContent_w(const MediaContentDescription* content,
1301 ContentAction action) {
1302 bool ret = UpdateRemoteStreams_w(content->streams(), action);
1303 // Set remote SRTP parameters (what the other side will encrypt with).
1304 ret &= SetSrtp_w(content->cryptos(), action, CS_REMOTE);
1305 // Set remote RTCP mux parameters.
1306 ret &= SetRtcpMux_w(content->rtcp_mux(), action, CS_REMOTE);
1307 // Set remote RTP header extensions.
1308 if (content->rtp_header_extensions_set()) {
1309 ret &= media_channel()->SetSendRtpHeaderExtensions(
1310 content->rtp_header_extensions());
1311 }
1312 if (content->bandwidth() != kAutoBandwidth) {
1313 ret &= media_channel()->SetSendBandwidth(false, content->bandwidth());
1314 }
1315 set_remote_content_direction(content->direction());
1316 return ret;
1317}
1318
1319void BaseChannel::OnMessage(talk_base::Message *pmsg) {
1320 switch (pmsg->message_id) {
1321 case MSG_ENABLE:
1322 EnableMedia_w();
1323 break;
1324 case MSG_DISABLE:
1325 DisableMedia_w();
1326 break;
1327 case MSG_MUTESTREAM: {
1328 MuteStreamData* data = static_cast<MuteStreamData*>(pmsg->pdata);
1329 data->result = MuteStream_w(data->ssrc, data->mute);
1330 break;
1331 }
1332 case MSG_ISSTREAMMUTED: {
1333 SsrcMessageData* data = static_cast<SsrcMessageData*>(pmsg->pdata);
1334 data->result = IsStreamMuted_w(data->ssrc);
1335 break;
1336 }
1337 case MSG_SETLOCALCONTENT: {
1338 SetContentData* data = static_cast<SetContentData*>(pmsg->pdata);
1339 data->result = SetLocalContent_w(data->content, data->action);
1340 break;
1341 }
1342 case MSG_SETREMOTECONTENT: {
1343 SetContentData* data = static_cast<SetContentData*>(pmsg->pdata);
1344 data->result = SetRemoteContent_w(data->content, data->action);
1345 break;
1346 }
1347 case MSG_ADDRECVSTREAM: {
1348 StreamMessageData* data = static_cast<StreamMessageData*>(pmsg->pdata);
1349 data->result = AddRecvStream_w(data->sp);
1350 break;
1351 }
1352 case MSG_REMOVERECVSTREAM: {
1353 SsrcMessageData* data = static_cast<SsrcMessageData*>(pmsg->pdata);
1354 data->result = RemoveRecvStream_w(data->ssrc);
1355 break;
1356 }
1357 case MSG_SETMAXSENDBANDWIDTH: {
1358 SetBandwidthData* data = static_cast<SetBandwidthData*>(pmsg->pdata);
1359 data->result = SetMaxSendBandwidth_w(data->value);
1360 break;
1361 }
1362
1363 case MSG_RTPPACKET:
1364 case MSG_RTCPPACKET: {
1365 PacketMessageData* data = static_cast<PacketMessageData*>(pmsg->pdata);
1366 SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet);
1367 delete data; // because it is Posted
1368 break;
1369 }
1370 case MSG_FIRSTPACKETRECEIVED: {
1371 SignalFirstPacketReceived(this);
1372 break;
1373 }
1374 case MSG_SESSION_ERROR: {
1375 SessionErrorMessageData* data = static_cast<SessionErrorMessageData*>
1376 (pmsg->pdata);
1377 session_->SetError(data->error_);
1378 break;
1379 }
1380 }
1381}
1382
1383void BaseChannel::Send(uint32 id, talk_base::MessageData *pdata) {
1384 worker_thread_->Send(this, id, pdata);
1385}
1386
1387void BaseChannel::Post(uint32 id, talk_base::MessageData *pdata) {
1388 worker_thread_->Post(this, id, pdata);
1389}
1390
1391void BaseChannel::PostDelayed(int cmsDelay, uint32 id,
1392 talk_base::MessageData *pdata) {
1393 worker_thread_->PostDelayed(cmsDelay, this, id, pdata);
1394}
1395
1396void BaseChannel::Clear(uint32 id, talk_base::MessageList* removed) {
1397 worker_thread_->Clear(this, id, removed);
1398}
1399
1400void BaseChannel::FlushRtcpMessages() {
1401 // Flush all remaining RTCP messages. This should only be called in
1402 // destructor.
1403 ASSERT(talk_base::Thread::Current() == worker_thread_);
1404 talk_base::MessageList rtcp_messages;
1405 Clear(MSG_RTCPPACKET, &rtcp_messages);
1406 for (talk_base::MessageList::iterator it = rtcp_messages.begin();
1407 it != rtcp_messages.end(); ++it) {
1408 Send(MSG_RTCPPACKET, it->pdata);
1409 }
1410}
1411
1412VoiceChannel::VoiceChannel(talk_base::Thread* thread,
1413 MediaEngineInterface* media_engine,
1414 VoiceMediaChannel* media_channel,
1415 BaseSession* session,
1416 const std::string& content_name,
1417 bool rtcp)
1418 : BaseChannel(thread, media_engine, media_channel, session, content_name,
1419 rtcp),
1420 received_media_(false) {
1421}
1422
1423VoiceChannel::~VoiceChannel() {
1424 StopAudioMonitor();
1425 StopMediaMonitor();
1426 // this can't be done in the base class, since it calls a virtual
1427 DisableMedia_w();
1428}
1429
1430bool VoiceChannel::Init() {
1431 TransportChannel* rtcp_channel = rtcp() ? session()->CreateChannel(
1432 content_name(), "rtcp", ICE_CANDIDATE_COMPONENT_RTCP) : NULL;
1433 if (!BaseChannel::Init(session()->CreateChannel(
1434 content_name(), "rtp", ICE_CANDIDATE_COMPONENT_RTP),
1435 rtcp_channel)) {
1436 return false;
1437 }
1438 media_channel()->SignalMediaError.connect(
1439 this, &VoiceChannel::OnVoiceChannelError);
1440 srtp_filter()->SignalSrtpError.connect(
1441 this, &VoiceChannel::OnSrtpError);
1442 return true;
1443}
1444
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00001445bool VoiceChannel::SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) {
1446 AudioRenderMessageData data(ssrc, renderer, false);
1447 Send(MSG_SETRENDERER, &data);
1448 return data.result;
1449}
1450
1451bool VoiceChannel::SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) {
1452 AudioRenderMessageData data(ssrc, renderer, true);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001453 Send(MSG_SETRENDERER, &data);
1454 return data.result;
1455}
1456
1457bool VoiceChannel::SetRingbackTone(const void* buf, int len) {
1458 SetRingbackToneMessageData data(buf, len);
1459 Send(MSG_SETRINGBACKTONE, &data);
1460 return data.result;
1461}
1462
1463// TODO(juberti): Handle early media the right way. We should get an explicit
1464// ringing message telling us to start playing local ringback, which we cancel
1465// if any early media actually arrives. For now, we do the opposite, which is
1466// to wait 1 second for early media, and start playing local ringback if none
1467// arrives.
1468void VoiceChannel::SetEarlyMedia(bool enable) {
1469 if (enable) {
1470 // Start the early media timeout
1471 PostDelayed(kEarlyMediaTimeout, MSG_EARLYMEDIATIMEOUT);
1472 } else {
1473 // Stop the timeout if currently going.
1474 Clear(MSG_EARLYMEDIATIMEOUT);
1475 }
1476}
1477
1478bool VoiceChannel::PlayRingbackTone(uint32 ssrc, bool play, bool loop) {
1479 PlayRingbackToneMessageData data(ssrc, play, loop);
1480 Send(MSG_PLAYRINGBACKTONE, &data);
1481 return data.result;
1482}
1483
1484bool VoiceChannel::PressDTMF(int digit, bool playout) {
1485 int flags = DF_SEND;
1486 if (playout) {
1487 flags |= DF_PLAY;
1488 }
1489 int duration_ms = 160;
1490 return InsertDtmf(0, digit, duration_ms, flags);
1491}
1492
1493bool VoiceChannel::CanInsertDtmf() {
1494 BoolMessageData data(false);
1495 Send(MSG_CANINSERTDTMF, &data);
1496 return data.data();
1497}
1498
1499bool VoiceChannel::InsertDtmf(uint32 ssrc, int event_code, int duration,
1500 int flags) {
1501 DtmfMessageData data(ssrc, event_code, duration, flags);
1502 Send(MSG_INSERTDTMF, &data);
1503 return data.result;
1504}
1505
1506bool VoiceChannel::SetOutputScaling(uint32 ssrc, double left, double right) {
1507 ScaleVolumeMessageData data(ssrc, left, right);
1508 Send(MSG_SCALEVOLUME, &data);
1509 return data.result;
1510}
1511bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
1512 VoiceStatsMessageData data(stats);
1513 Send(MSG_GETSTATS, &data);
1514 return data.result;
1515}
1516
1517void VoiceChannel::StartMediaMonitor(int cms) {
1518 media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(),
1519 talk_base::Thread::Current()));
1520 media_monitor_->SignalUpdate.connect(
1521 this, &VoiceChannel::OnMediaMonitorUpdate);
1522 media_monitor_->Start(cms);
1523}
1524
1525void VoiceChannel::StopMediaMonitor() {
1526 if (media_monitor_) {
1527 media_monitor_->Stop();
1528 media_monitor_->SignalUpdate.disconnect(this);
1529 media_monitor_.reset();
1530 }
1531}
1532
1533void VoiceChannel::StartAudioMonitor(int cms) {
1534 audio_monitor_.reset(new AudioMonitor(this, talk_base::Thread::Current()));
1535 audio_monitor_
1536 ->SignalUpdate.connect(this, &VoiceChannel::OnAudioMonitorUpdate);
1537 audio_monitor_->Start(cms);
1538}
1539
1540void VoiceChannel::StopAudioMonitor() {
1541 if (audio_monitor_) {
1542 audio_monitor_->Stop();
1543 audio_monitor_.reset();
1544 }
1545}
1546
1547bool VoiceChannel::IsAudioMonitorRunning() const {
1548 return (audio_monitor_.get() != NULL);
1549}
1550
1551void VoiceChannel::StartTypingMonitor(const TypingMonitorOptions& settings) {
1552 typing_monitor_.reset(new TypingMonitor(this, worker_thread(), settings));
1553 SignalAutoMuted.repeat(typing_monitor_->SignalMuted);
1554}
1555
1556void VoiceChannel::StopTypingMonitor() {
1557 typing_monitor_.reset();
1558}
1559
1560bool VoiceChannel::IsTypingMonitorRunning() const {
1561 return typing_monitor_;
1562}
1563
1564bool VoiceChannel::MuteStream_w(uint32 ssrc, bool mute) {
1565 bool ret = BaseChannel::MuteStream_w(ssrc, mute);
1566 if (typing_monitor_ && mute)
1567 typing_monitor_->OnChannelMuted();
1568 return ret;
1569}
1570
1571int VoiceChannel::GetInputLevel_w() {
1572 return media_engine()->GetInputLevel();
1573}
1574
1575int VoiceChannel::GetOutputLevel_w() {
1576 return media_channel()->GetOutputLevel();
1577}
1578
1579void VoiceChannel::GetActiveStreams_w(AudioInfo::StreamList* actives) {
1580 media_channel()->GetActiveStreams(actives);
1581}
1582
1583void VoiceChannel::OnChannelRead(TransportChannel* channel,
1584 const char* data, size_t len, int flags) {
1585 BaseChannel::OnChannelRead(channel, data, len, flags);
1586
1587 // Set a flag when we've received an RTP packet. If we're waiting for early
1588 // media, this will disable the timeout.
1589 if (!received_media_ && !PacketIsRtcp(channel, data, len)) {
1590 received_media_ = true;
1591 }
1592}
1593
1594void VoiceChannel::ChangeState() {
1595 // Render incoming data if we're the active call, and we have the local
1596 // content. We receive data on the default channel and multiplexed streams.
1597 bool recv = IsReadyToReceive();
1598 if (!media_channel()->SetPlayout(recv)) {
1599 SendLastMediaError();
1600 }
1601
1602 // Send outgoing data if we're the active call, we have the remote content,
1603 // and we have had some form of connectivity.
1604 bool send = IsReadyToSend();
1605 SendFlags send_flag = send ? SEND_MICROPHONE : SEND_NOTHING;
1606 if (!media_channel()->SetSend(send_flag)) {
1607 LOG(LS_ERROR) << "Failed to SetSend " << send_flag << " on voice channel";
1608 SendLastMediaError();
1609 }
1610
1611 LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send;
1612}
1613
1614const ContentInfo* VoiceChannel::GetFirstContent(
1615 const SessionDescription* sdesc) {
1616 return GetFirstAudioContent(sdesc);
1617}
1618
1619bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
1620 ContentAction action) {
1621 ASSERT(worker_thread() == talk_base::Thread::Current());
1622 LOG(LS_INFO) << "Setting local voice description";
1623
1624 const AudioContentDescription* audio =
1625 static_cast<const AudioContentDescription*>(content);
1626 ASSERT(audio != NULL);
1627 if (!audio) return false;
1628
1629 bool ret = SetBaseLocalContent_w(content, action);
1630 // Set local audio codecs (what we want to receive).
1631 // TODO(whyuan): Change action != CA_UPDATE to !audio->partial() when partial
1632 // is set properly.
1633 if (action != CA_UPDATE || audio->has_codecs()) {
1634 ret &= media_channel()->SetRecvCodecs(audio->codecs());
1635 }
1636
1637 // If everything worked, see if we can start receiving.
1638 if (ret) {
1639 ChangeState();
1640 } else {
1641 LOG(LS_WARNING) << "Failed to set local voice description";
1642 }
1643 return ret;
1644}
1645
1646bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
1647 ContentAction action) {
1648 ASSERT(worker_thread() == talk_base::Thread::Current());
1649 LOG(LS_INFO) << "Setting remote voice description";
1650
1651 const AudioContentDescription* audio =
1652 static_cast<const AudioContentDescription*>(content);
1653 ASSERT(audio != NULL);
1654 if (!audio) return false;
1655
1656 bool ret = true;
1657 // Set remote video codecs (what the other side wants to receive).
1658 if (action != CA_UPDATE || audio->has_codecs()) {
1659 ret &= media_channel()->SetSendCodecs(audio->codecs());
1660 }
1661
1662 ret &= SetBaseRemoteContent_w(content, action);
1663
1664 if (action != CA_UPDATE) {
1665 // Tweak our audio processing settings, if needed.
1666 AudioOptions audio_options;
1667 if (!media_channel()->GetOptions(&audio_options)) {
1668 LOG(LS_WARNING) << "Can not set audio options from on remote content.";
1669 } else {
1670 if (audio->conference_mode()) {
1671 audio_options.conference_mode.Set(true);
1672 }
1673 if (audio->agc_minus_10db()) {
1674 audio_options.adjust_agc_delta.Set(kAgcMinus10db);
1675 }
1676 if (!media_channel()->SetOptions(audio_options)) {
1677 // Log an error on failure, but don't abort the call.
1678 LOG(LS_ERROR) << "Failed to set voice channel options";
1679 }
1680 }
1681 }
1682
1683 // If everything worked, see if we can start sending.
1684 if (ret) {
1685 ChangeState();
1686 } else {
1687 LOG(LS_WARNING) << "Failed to set remote voice description";
1688 }
1689 return ret;
1690}
1691
1692bool VoiceChannel::SetRingbackTone_w(const void* buf, int len) {
1693 ASSERT(worker_thread() == talk_base::Thread::Current());
1694 return media_channel()->SetRingbackTone(static_cast<const char*>(buf), len);
1695}
1696
1697bool VoiceChannel::PlayRingbackTone_w(uint32 ssrc, bool play, bool loop) {
1698 ASSERT(worker_thread() == talk_base::Thread::Current());
1699 if (play) {
1700 LOG(LS_INFO) << "Playing ringback tone, loop=" << loop;
1701 } else {
1702 LOG(LS_INFO) << "Stopping ringback tone";
1703 }
1704 return media_channel()->PlayRingbackTone(ssrc, play, loop);
1705}
1706
1707void VoiceChannel::HandleEarlyMediaTimeout() {
1708 // This occurs on the main thread, not the worker thread.
1709 if (!received_media_) {
1710 LOG(LS_INFO) << "No early media received before timeout";
1711 SignalEarlyMediaTimeout(this);
1712 }
1713}
1714
1715bool VoiceChannel::CanInsertDtmf_w() {
1716 return media_channel()->CanInsertDtmf();
1717}
1718
1719bool VoiceChannel::InsertDtmf_w(uint32 ssrc, int event, int duration,
1720 int flags) {
1721 if (!enabled()) {
1722 return false;
1723 }
1724
1725 return media_channel()->InsertDtmf(ssrc, event, duration, flags);
1726}
1727
1728bool VoiceChannel::SetOutputScaling_w(uint32 ssrc, double left, double right) {
1729 return media_channel()->SetOutputScaling(ssrc, left, right);
1730}
1731
1732bool VoiceChannel::GetStats_w(VoiceMediaInfo* stats) {
1733 return media_channel()->GetStats(stats);
1734}
1735
1736bool VoiceChannel::SetChannelOptions(const AudioOptions& options) {
1737 AudioOptionsMessageData data(options);
1738 Send(MSG_SETCHANNELOPTIONS, &data);
1739 return data.result;
1740}
1741
1742bool VoiceChannel::SetChannelOptions_w(const AudioOptions& options) {
1743 return media_channel()->SetOptions(options);
1744}
1745
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00001746bool VoiceChannel::SetRenderer_w(uint32 ssrc, AudioRenderer* renderer,
1747 bool is_local) {
1748 if (is_local)
1749 return media_channel()->SetLocalRenderer(ssrc, renderer);
1750
1751 return media_channel()->SetRemoteRenderer(ssrc, renderer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001752}
1753
1754void VoiceChannel::OnMessage(talk_base::Message *pmsg) {
1755 switch (pmsg->message_id) {
1756 case MSG_SETRINGBACKTONE: {
1757 SetRingbackToneMessageData* data =
1758 static_cast<SetRingbackToneMessageData*>(pmsg->pdata);
1759 data->result = SetRingbackTone_w(data->buf, data->len);
1760 break;
1761 }
1762 case MSG_PLAYRINGBACKTONE: {
1763 PlayRingbackToneMessageData* data =
1764 static_cast<PlayRingbackToneMessageData*>(pmsg->pdata);
1765 data->result = PlayRingbackTone_w(data->ssrc, data->play, data->loop);
1766 break;
1767 }
1768 case MSG_EARLYMEDIATIMEOUT:
1769 HandleEarlyMediaTimeout();
1770 break;
1771 case MSG_CANINSERTDTMF: {
1772 BoolMessageData* data =
1773 static_cast<BoolMessageData*>(pmsg->pdata);
1774 data->data() = CanInsertDtmf_w();
1775 break;
1776 }
1777 case MSG_INSERTDTMF: {
1778 DtmfMessageData* data =
1779 static_cast<DtmfMessageData*>(pmsg->pdata);
1780 data->result = InsertDtmf_w(data->ssrc, data->event, data->duration,
1781 data->flags);
1782 break;
1783 }
1784 case MSG_SCALEVOLUME: {
1785 ScaleVolumeMessageData* data =
1786 static_cast<ScaleVolumeMessageData*>(pmsg->pdata);
1787 data->result = SetOutputScaling_w(data->ssrc, data->left, data->right);
1788 break;
1789 }
1790 case MSG_GETSTATS: {
1791 VoiceStatsMessageData* data =
1792 static_cast<VoiceStatsMessageData*>(pmsg->pdata);
1793 data->result = GetStats_w(data->stats);
1794 break;
1795 }
1796 case MSG_CHANNEL_ERROR: {
1797 VoiceChannelErrorMessageData* data =
1798 static_cast<VoiceChannelErrorMessageData*>(pmsg->pdata);
1799 SignalMediaError(this, data->ssrc, data->error);
1800 delete data;
1801 break;
1802 }
1803 case MSG_SETCHANNELOPTIONS: {
1804 AudioOptionsMessageData* data =
1805 static_cast<AudioOptionsMessageData*>(pmsg->pdata);
1806 data->result = SetChannelOptions_w(data->options);
1807 break;
1808 }
1809 case MSG_SETRENDERER: {
1810 AudioRenderMessageData* data =
1811 static_cast<AudioRenderMessageData*>(pmsg->pdata);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00001812 data->result = SetRenderer_w(data->ssrc, data->renderer, data->is_local);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001813 break;
1814 }
1815 default:
1816 BaseChannel::OnMessage(pmsg);
1817 break;
1818 }
1819}
1820
1821void VoiceChannel::OnConnectionMonitorUpdate(
1822 SocketMonitor* monitor, const std::vector<ConnectionInfo>& infos) {
1823 SignalConnectionMonitor(this, infos);
1824}
1825
1826void VoiceChannel::OnMediaMonitorUpdate(
1827 VoiceMediaChannel* media_channel, const VoiceMediaInfo& info) {
1828 ASSERT(media_channel == this->media_channel());
1829 SignalMediaMonitor(this, info);
1830}
1831
1832void VoiceChannel::OnAudioMonitorUpdate(AudioMonitor* monitor,
1833 const AudioInfo& info) {
1834 SignalAudioMonitor(this, info);
1835}
1836
1837void VoiceChannel::OnVoiceChannelError(
1838 uint32 ssrc, VoiceMediaChannel::Error err) {
1839 VoiceChannelErrorMessageData* data = new VoiceChannelErrorMessageData(
1840 ssrc, err);
1841 signaling_thread()->Post(this, MSG_CHANNEL_ERROR, data);
1842}
1843
1844void VoiceChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode,
1845 SrtpFilter::Error error) {
1846 switch (error) {
1847 case SrtpFilter::ERROR_FAIL:
1848 OnVoiceChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
1849 VoiceMediaChannel::ERROR_REC_SRTP_ERROR :
1850 VoiceMediaChannel::ERROR_PLAY_SRTP_ERROR);
1851 break;
1852 case SrtpFilter::ERROR_AUTH:
1853 OnVoiceChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
1854 VoiceMediaChannel::ERROR_REC_SRTP_AUTH_FAILED :
1855 VoiceMediaChannel::ERROR_PLAY_SRTP_AUTH_FAILED);
1856 break;
1857 case SrtpFilter::ERROR_REPLAY:
1858 // Only receving channel should have this error.
1859 ASSERT(mode == SrtpFilter::UNPROTECT);
1860 OnVoiceChannelError(ssrc, VoiceMediaChannel::ERROR_PLAY_SRTP_REPLAY);
1861 break;
1862 default:
1863 break;
1864 }
1865}
1866
1867void VoiceChannel::GetSrtpCiphers(std::vector<std::string>* ciphers) const {
1868 GetSupportedAudioCryptoSuites(ciphers);
1869}
1870
1871VideoChannel::VideoChannel(talk_base::Thread* thread,
1872 MediaEngineInterface* media_engine,
1873 VideoMediaChannel* media_channel,
1874 BaseSession* session,
1875 const std::string& content_name,
1876 bool rtcp,
1877 VoiceChannel* voice_channel)
1878 : BaseChannel(thread, media_engine, media_channel, session, content_name,
1879 rtcp),
1880 voice_channel_(voice_channel),
1881 renderer_(NULL),
1882 screencapture_factory_(CreateScreenCapturerFactory()),
1883 previous_we_(talk_base::WE_CLOSE) {
1884}
1885
1886bool VideoChannel::Init() {
1887 TransportChannel* rtcp_channel = rtcp() ? session()->CreateChannel(
1888 content_name(), "video_rtcp", ICE_CANDIDATE_COMPONENT_RTCP) : NULL;
1889 if (!BaseChannel::Init(session()->CreateChannel(
1890 content_name(), "video_rtp", ICE_CANDIDATE_COMPONENT_RTP),
1891 rtcp_channel)) {
1892 return false;
1893 }
1894 media_channel()->SignalMediaError.connect(
1895 this, &VideoChannel::OnVideoChannelError);
1896 srtp_filter()->SignalSrtpError.connect(
1897 this, &VideoChannel::OnSrtpError);
1898 return true;
1899}
1900
1901void VoiceChannel::SendLastMediaError() {
1902 uint32 ssrc;
1903 VoiceMediaChannel::Error error;
1904 media_channel()->GetLastMediaError(&ssrc, &error);
1905 SignalMediaError(this, ssrc, error);
1906}
1907
1908VideoChannel::~VideoChannel() {
1909 std::vector<uint32> screencast_ssrcs;
1910 ScreencastMap::iterator iter;
1911 while (!screencast_capturers_.empty()) {
1912 if (!RemoveScreencast(screencast_capturers_.begin()->first)) {
1913 LOG(LS_ERROR) << "Unable to delete screencast with ssrc "
1914 << screencast_capturers_.begin()->first;
1915 ASSERT(false);
1916 break;
1917 }
1918 }
1919
1920 StopMediaMonitor();
1921 // this can't be done in the base class, since it calls a virtual
1922 DisableMedia_w();
1923}
1924
1925bool VideoChannel::SetRenderer(uint32 ssrc, VideoRenderer* renderer) {
1926 VideoRenderMessageData data(ssrc, renderer);
1927 Send(MSG_SETRENDERER, &data);
1928 return true;
1929}
1930
1931bool VideoChannel::ApplyViewRequest(const ViewRequest& request) {
1932 ViewRequestMessageData data(request);
1933 Send(MSG_HANDLEVIEWREQUEST, &data);
1934 return data.result;
1935}
1936
1937VideoCapturer* VideoChannel::AddScreencast(
1938 uint32 ssrc, const ScreencastId& id) {
1939 AddScreencastMessageData data(ssrc, id);
1940 Send(MSG_ADDSCREENCAST, &data);
1941 return data.result;
1942}
1943
1944bool VideoChannel::SetCapturer(uint32 ssrc, VideoCapturer* capturer) {
1945 SetCapturerMessageData data(ssrc, capturer);
1946 Send(MSG_SETCAPTURER, &data);
1947 return data.result;
1948}
1949
1950bool VideoChannel::RemoveScreencast(uint32 ssrc) {
1951 RemoveScreencastMessageData data(ssrc);
1952 Send(MSG_REMOVESCREENCAST, &data);
1953 return data.result;
1954}
1955
1956bool VideoChannel::IsScreencasting() {
1957 IsScreencastingMessageData data;
1958 Send(MSG_ISSCREENCASTING, &data);
1959 return data.result;
1960}
1961
1962int VideoChannel::ScreencastFps(uint32 ssrc) {
1963 ScreencastFpsMessageData data(ssrc);
1964 Send(MSG_SCREENCASTFPS, &data);
1965 return data.result;
1966}
1967
1968bool VideoChannel::SendIntraFrame() {
1969 Send(MSG_SENDINTRAFRAME);
1970 return true;
1971}
1972
1973bool VideoChannel::RequestIntraFrame() {
1974 Send(MSG_REQUESTINTRAFRAME);
1975 return true;
1976}
1977
1978void VideoChannel::SetScreenCaptureFactory(
1979 ScreenCapturerFactory* screencapture_factory) {
1980 SetScreenCaptureFactoryMessageData data(screencapture_factory);
1981 Send(MSG_SETSCREENCASTFACTORY, &data);
1982}
1983
1984void VideoChannel::ChangeState() {
1985 // Render incoming data if we're the active call, and we have the local
1986 // content. We receive data on the default channel and multiplexed streams.
1987 bool recv = IsReadyToReceive();
1988 if (!media_channel()->SetRender(recv)) {
1989 LOG(LS_ERROR) << "Failed to SetRender on video channel";
1990 // TODO(gangji): Report error back to server.
1991 }
1992
1993 // Send outgoing data if we're the active call, we have the remote content,
1994 // and we have had some form of connectivity.
1995 bool send = IsReadyToSend();
1996 if (!media_channel()->SetSend(send)) {
1997 LOG(LS_ERROR) << "Failed to SetSend on video channel";
1998 // TODO(gangji): Report error back to server.
1999 }
2000
2001 LOG(LS_INFO) << "Changing video state, recv=" << recv << " send=" << send;
2002}
2003
2004bool VideoChannel::GetStats(VideoMediaInfo* stats) {
2005 VideoStatsMessageData data(stats);
2006 Send(MSG_GETSTATS, &data);
2007 return data.result;
2008}
2009
2010void VideoChannel::StartMediaMonitor(int cms) {
2011 media_monitor_.reset(new VideoMediaMonitor(media_channel(), worker_thread(),
2012 talk_base::Thread::Current()));
2013 media_monitor_->SignalUpdate.connect(
2014 this, &VideoChannel::OnMediaMonitorUpdate);
2015 media_monitor_->Start(cms);
2016}
2017
2018void VideoChannel::StopMediaMonitor() {
2019 if (media_monitor_) {
2020 media_monitor_->Stop();
2021 media_monitor_.reset();
2022 }
2023}
2024
2025const ContentInfo* VideoChannel::GetFirstContent(
2026 const SessionDescription* sdesc) {
2027 return GetFirstVideoContent(sdesc);
2028}
2029
2030bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
2031 ContentAction action) {
2032 ASSERT(worker_thread() == talk_base::Thread::Current());
2033 LOG(LS_INFO) << "Setting local video description";
2034
2035 const VideoContentDescription* video =
2036 static_cast<const VideoContentDescription*>(content);
2037 ASSERT(video != NULL);
2038 if (!video) return false;
2039
2040 bool ret = SetBaseLocalContent_w(content, action);
2041 // Set local video codecs (what we want to receive).
2042 if (action != CA_UPDATE || video->has_codecs()) {
2043 ret &= media_channel()->SetRecvCodecs(video->codecs());
2044 }
2045
2046 if (action != CA_UPDATE) {
2047 VideoOptions video_options;
2048 media_channel()->GetOptions(&video_options);
2049 video_options.buffered_mode_latency.Set(video->buffered_mode_latency());
2050
2051 if (!media_channel()->SetOptions(video_options)) {
2052 // Log an error on failure, but don't abort the call.
2053 LOG(LS_ERROR) << "Failed to set video channel options";
2054 }
2055 }
2056
2057 // If everything worked, see if we can start receiving.
2058 if (ret) {
2059 ChangeState();
2060 } else {
2061 LOG(LS_WARNING) << "Failed to set local video description";
2062 }
2063 return ret;
2064}
2065
2066bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
2067 ContentAction action) {
2068 ASSERT(worker_thread() == talk_base::Thread::Current());
2069 LOG(LS_INFO) << "Setting remote video description";
2070
2071 const VideoContentDescription* video =
2072 static_cast<const VideoContentDescription*>(content);
2073 ASSERT(video != NULL);
2074 if (!video) return false;
2075
2076 bool ret = true;
2077 // Set remote video codecs (what the other side wants to receive).
2078 if (action != CA_UPDATE || video->has_codecs()) {
2079 ret &= media_channel()->SetSendCodecs(video->codecs());
2080 }
2081
2082 ret &= SetBaseRemoteContent_w(content, action);
2083
2084 if (action != CA_UPDATE) {
2085 // Tweak our video processing settings, if needed.
2086 VideoOptions video_options;
2087 media_channel()->GetOptions(&video_options);
2088 video_options.conference_mode.Set(video->conference_mode());
2089 video_options.buffered_mode_latency.Set(video->buffered_mode_latency());
2090
2091 if (!media_channel()->SetOptions(video_options)) {
2092 // Log an error on failure, but don't abort the call.
2093 LOG(LS_ERROR) << "Failed to set video channel options";
2094 }
2095 }
2096
2097 // If everything worked, see if we can start sending.
2098 if (ret) {
2099 ChangeState();
2100 } else {
2101 LOG(LS_WARNING) << "Failed to set remote video description";
2102 }
2103 return ret;
2104}
2105
2106bool VideoChannel::ApplyViewRequest_w(const ViewRequest& request) {
2107 bool ret = true;
2108 // Set the send format for each of the local streams. If the view request
2109 // does not contain a local stream, set its send format to 0x0, which will
2110 // drop all frames.
2111 for (std::vector<StreamParams>::const_iterator it = local_streams().begin();
2112 it != local_streams().end(); ++it) {
2113 VideoFormat format(0, 0, 0, cricket::FOURCC_I420);
2114 StaticVideoViews::const_iterator view;
2115 for (view = request.static_video_views.begin();
2116 view != request.static_video_views.end(); ++view) {
2117 if (view->selector.Matches(*it)) {
2118 format.width = view->width;
2119 format.height = view->height;
2120 format.interval = cricket::VideoFormat::FpsToInterval(view->framerate);
2121 break;
2122 }
2123 }
2124
2125 ret &= media_channel()->SetSendStreamFormat(it->first_ssrc(), format);
2126 }
2127
2128 // Check if the view request has invalid streams.
2129 for (StaticVideoViews::const_iterator it = request.static_video_views.begin();
2130 it != request.static_video_views.end(); ++it) {
2131 if (!GetStream(local_streams(), it->selector, NULL)) {
2132 LOG(LS_WARNING) << "View request for ("
2133 << it->selector.ssrc << ", '"
2134 << it->selector.groupid << "', '"
2135 << it->selector.streamid << "'"
2136 << ") is not in the local streams.";
2137 }
2138 }
2139
2140 return ret;
2141}
2142
2143void VideoChannel::SetRenderer_w(uint32 ssrc, VideoRenderer* renderer) {
2144 media_channel()->SetRenderer(ssrc, renderer);
2145}
2146
2147VideoCapturer* VideoChannel::AddScreencast_w(
2148 uint32 ssrc, const ScreencastId& id) {
2149 if (screencast_capturers_.find(ssrc) != screencast_capturers_.end()) {
2150 return NULL;
2151 }
2152 VideoCapturer* screen_capturer =
2153 screencapture_factory_->CreateScreenCapturer(id);
2154 if (!screen_capturer) {
2155 return NULL;
2156 }
2157 screen_capturer->SignalStateChange.connect(this,
2158 &VideoChannel::OnStateChange);
2159 screencast_capturers_[ssrc] = screen_capturer;
2160 return screen_capturer;
2161}
2162
2163bool VideoChannel::SetCapturer_w(uint32 ssrc, VideoCapturer* capturer) {
2164 return media_channel()->SetCapturer(ssrc, capturer);
2165}
2166
2167bool VideoChannel::RemoveScreencast_w(uint32 ssrc) {
2168 ScreencastMap::iterator iter = screencast_capturers_.find(ssrc);
2169 if (iter == screencast_capturers_.end()) {
2170 return false;
2171 }
2172 // Clean up VideoCapturer.
2173 delete iter->second;
2174 screencast_capturers_.erase(iter);
2175 return true;
2176}
2177
2178bool VideoChannel::IsScreencasting_w() const {
2179 return !screencast_capturers_.empty();
2180}
2181
2182int VideoChannel::ScreencastFps_w(uint32 ssrc) const {
2183 ScreencastMap::const_iterator iter = screencast_capturers_.find(ssrc);
2184 if (iter == screencast_capturers_.end()) {
2185 return 0;
2186 }
2187 VideoCapturer* capturer = iter->second;
2188 const VideoFormat* video_format = capturer->GetCaptureFormat();
2189 return VideoFormat::IntervalToFps(video_format->interval);
2190}
2191
2192void VideoChannel::SetScreenCaptureFactory_w(
2193 ScreenCapturerFactory* screencapture_factory) {
2194 if (screencapture_factory == NULL) {
2195 screencapture_factory_.reset(CreateScreenCapturerFactory());
2196 } else {
2197 screencapture_factory_.reset(screencapture_factory);
2198 }
2199}
2200
2201bool VideoChannel::GetStats_w(VideoMediaInfo* stats) {
2202 return media_channel()->GetStats(stats);
2203}
2204
2205void VideoChannel::OnScreencastWindowEvent_s(uint32 ssrc,
2206 talk_base::WindowEvent we) {
2207 ASSERT(signaling_thread() == talk_base::Thread::Current());
2208 SignalScreencastWindowEvent(ssrc, we);
2209}
2210
2211bool VideoChannel::SetChannelOptions(const VideoOptions &options) {
2212 VideoOptionsMessageData data(options);
2213 Send(MSG_SETCHANNELOPTIONS, &data);
2214 return data.result;
2215}
2216
2217bool VideoChannel::SetChannelOptions_w(const VideoOptions &options) {
2218 return media_channel()->SetOptions(options);
2219}
2220
2221void VideoChannel::OnMessage(talk_base::Message *pmsg) {
2222 switch (pmsg->message_id) {
2223 case MSG_SETRENDERER: {
2224 const VideoRenderMessageData* data =
2225 static_cast<VideoRenderMessageData*>(pmsg->pdata);
2226 SetRenderer_w(data->ssrc, data->renderer);
2227 break;
2228 }
2229 case MSG_ADDSCREENCAST: {
2230 AddScreencastMessageData* data =
2231 static_cast<AddScreencastMessageData*>(pmsg->pdata);
2232 data->result = AddScreencast_w(data->ssrc, data->window_id);
2233 break;
2234 }
2235 case MSG_SETCAPTURER: {
2236 SetCapturerMessageData* data =
2237 static_cast<SetCapturerMessageData*>(pmsg->pdata);
2238 data->result = SetCapturer_w(data->ssrc, data->capturer);
2239 break;
2240 }
2241 case MSG_REMOVESCREENCAST: {
2242 RemoveScreencastMessageData* data =
2243 static_cast<RemoveScreencastMessageData*>(pmsg->pdata);
2244 data->result = RemoveScreencast_w(data->ssrc);
2245 break;
2246 }
2247 case MSG_SCREENCASTWINDOWEVENT: {
2248 const ScreencastEventMessageData* data =
2249 static_cast<ScreencastEventMessageData*>(pmsg->pdata);
2250 OnScreencastWindowEvent_s(data->ssrc, data->event);
2251 delete data;
2252 break;
2253 }
2254 case MSG_ISSCREENCASTING: {
2255 IsScreencastingMessageData* data =
2256 static_cast<IsScreencastingMessageData*>(pmsg->pdata);
2257 data->result = IsScreencasting_w();
2258 break;
2259 }
2260 case MSG_SCREENCASTFPS: {
2261 ScreencastFpsMessageData* data =
2262 static_cast<ScreencastFpsMessageData*>(pmsg->pdata);
2263 data->result = ScreencastFps_w(data->ssrc);
2264 break;
2265 }
2266 case MSG_SENDINTRAFRAME: {
2267 SendIntraFrame_w();
2268 break;
2269 }
2270 case MSG_REQUESTINTRAFRAME: {
2271 RequestIntraFrame_w();
2272 break;
2273 }
2274 case MSG_SETCHANNELOPTIONS: {
2275 VideoOptionsMessageData* data =
2276 static_cast<VideoOptionsMessageData*>(pmsg->pdata);
2277 data->result = SetChannelOptions_w(data->options);
2278 break;
2279 }
2280 case MSG_CHANNEL_ERROR: {
2281 const VideoChannelErrorMessageData* data =
2282 static_cast<VideoChannelErrorMessageData*>(pmsg->pdata);
2283 SignalMediaError(this, data->ssrc, data->error);
2284 delete data;
2285 break;
2286 }
2287 case MSG_HANDLEVIEWREQUEST: {
2288 ViewRequestMessageData* data =
2289 static_cast<ViewRequestMessageData*>(pmsg->pdata);
2290 data->result = ApplyViewRequest_w(data->request);
2291 break;
2292 }
2293 case MSG_SETSCREENCASTFACTORY: {
2294 SetScreenCaptureFactoryMessageData* data =
2295 static_cast<SetScreenCaptureFactoryMessageData*>(pmsg->pdata);
2296 SetScreenCaptureFactory_w(data->screencapture_factory);
wu@webrtc.orgd64719d2013-08-01 00:00:07 +00002297 break;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002298 }
2299 case MSG_GETSTATS: {
2300 VideoStatsMessageData* data =
2301 static_cast<VideoStatsMessageData*>(pmsg->pdata);
2302 data->result = GetStats_w(data->stats);
2303 break;
2304 }
2305 default:
2306 BaseChannel::OnMessage(pmsg);
2307 break;
2308 }
2309}
2310
2311void VideoChannel::OnConnectionMonitorUpdate(
2312 SocketMonitor *monitor, const std::vector<ConnectionInfo> &infos) {
2313 SignalConnectionMonitor(this, infos);
2314}
2315
2316// TODO(pthatcher): Look into removing duplicate code between
2317// audio, video, and data, perhaps by using templates.
2318void VideoChannel::OnMediaMonitorUpdate(
2319 VideoMediaChannel* media_channel, const VideoMediaInfo &info) {
2320 ASSERT(media_channel == this->media_channel());
2321 SignalMediaMonitor(this, info);
2322}
2323
2324void VideoChannel::OnScreencastWindowEvent(uint32 ssrc,
2325 talk_base::WindowEvent event) {
2326 ScreencastEventMessageData* pdata =
2327 new ScreencastEventMessageData(ssrc, event);
2328 signaling_thread()->Post(this, MSG_SCREENCASTWINDOWEVENT, pdata);
2329}
2330
2331void VideoChannel::OnStateChange(VideoCapturer* capturer, CaptureState ev) {
2332 // Map capturer events to window events. In the future we may want to simply
2333 // pass these events up directly.
2334 talk_base::WindowEvent we;
2335 if (ev == CS_STOPPED) {
2336 we = talk_base::WE_CLOSE;
2337 } else if (ev == CS_PAUSED) {
2338 we = talk_base::WE_MINIMIZE;
2339 } else if (ev == CS_RUNNING && previous_we_ == talk_base::WE_MINIMIZE) {
2340 we = talk_base::WE_RESTORE;
2341 } else {
2342 return;
2343 }
2344 previous_we_ = we;
2345
2346 uint32 ssrc = 0;
2347 if (!GetLocalSsrc(capturer, &ssrc)) {
2348 return;
2349 }
2350 ScreencastEventMessageData* pdata =
2351 new ScreencastEventMessageData(ssrc, we);
2352 signaling_thread()->Post(this, MSG_SCREENCASTWINDOWEVENT, pdata);
2353}
2354
2355bool VideoChannel::GetLocalSsrc(const VideoCapturer* capturer, uint32* ssrc) {
2356 *ssrc = 0;
2357 for (ScreencastMap::iterator iter = screencast_capturers_.begin();
2358 iter != screencast_capturers_.end(); ++iter) {
2359 if (iter->second == capturer) {
2360 *ssrc = iter->first;
2361 return true;
2362 }
2363 }
2364 return false;
2365}
2366
2367void VideoChannel::OnVideoChannelError(uint32 ssrc,
2368 VideoMediaChannel::Error error) {
2369 VideoChannelErrorMessageData* data = new VideoChannelErrorMessageData(
2370 ssrc, error);
2371 signaling_thread()->Post(this, MSG_CHANNEL_ERROR, data);
2372}
2373
2374void VideoChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode,
2375 SrtpFilter::Error error) {
2376 switch (error) {
2377 case SrtpFilter::ERROR_FAIL:
2378 OnVideoChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
2379 VideoMediaChannel::ERROR_REC_SRTP_ERROR :
2380 VideoMediaChannel::ERROR_PLAY_SRTP_ERROR);
2381 break;
2382 case SrtpFilter::ERROR_AUTH:
2383 OnVideoChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
2384 VideoMediaChannel::ERROR_REC_SRTP_AUTH_FAILED :
2385 VideoMediaChannel::ERROR_PLAY_SRTP_AUTH_FAILED);
2386 break;
2387 case SrtpFilter::ERROR_REPLAY:
2388 // Only receving channel should have this error.
2389 ASSERT(mode == SrtpFilter::UNPROTECT);
2390 // TODO(gangji): Turn on the signaling of replay error once we have
2391 // switched to the new mechanism for doing video retransmissions.
2392 // OnVideoChannelError(ssrc, VideoMediaChannel::ERROR_PLAY_SRTP_REPLAY);
2393 break;
2394 default:
2395 break;
2396 }
2397}
2398
2399
2400void VideoChannel::GetSrtpCiphers(std::vector<std::string>* ciphers) const {
2401 GetSupportedVideoCryptoSuites(ciphers);
2402}
2403
2404DataChannel::DataChannel(talk_base::Thread* thread,
2405 DataMediaChannel* media_channel,
2406 BaseSession* session,
2407 const std::string& content_name,
2408 bool rtcp)
2409 // MediaEngine is NULL
2410 : BaseChannel(thread, NULL, media_channel, session, content_name, rtcp),
2411 data_channel_type_(cricket::DCT_NONE) {
2412}
2413
2414DataChannel::~DataChannel() {
2415 StopMediaMonitor();
2416 // this can't be done in the base class, since it calls a virtual
2417 DisableMedia_w();
2418}
2419
2420bool DataChannel::Init() {
2421 TransportChannel* rtcp_channel = rtcp() ? session()->CreateChannel(
2422 content_name(), "data_rtcp", ICE_CANDIDATE_COMPONENT_RTCP) : NULL;
2423 if (!BaseChannel::Init(session()->CreateChannel(
2424 content_name(), "data_rtp", ICE_CANDIDATE_COMPONENT_RTP),
2425 rtcp_channel)) {
2426 return false;
2427 }
2428 media_channel()->SignalDataReceived.connect(
2429 this, &DataChannel::OnDataReceived);
2430 media_channel()->SignalMediaError.connect(
2431 this, &DataChannel::OnDataChannelError);
wu@webrtc.orgd64719d2013-08-01 00:00:07 +00002432 media_channel()->SignalReadyToSend.connect(
2433 this, &DataChannel::OnDataChannelReadyToSend);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002434 srtp_filter()->SignalSrtpError.connect(
2435 this, &DataChannel::OnSrtpError);
2436 return true;
2437}
2438
2439bool DataChannel::SendData(const SendDataParams& params,
2440 const talk_base::Buffer& payload,
2441 SendDataResult* result) {
2442 SendDataMessageData message_data(params, &payload, result);
2443 Send(MSG_SENDDATA, &message_data);
2444 return message_data.succeeded;
2445}
2446
2447const ContentInfo* DataChannel::GetFirstContent(
2448 const SessionDescription* sdesc) {
2449 return GetFirstDataContent(sdesc);
2450}
2451
2452
2453static bool IsRtpPacket(const talk_base::Buffer* packet) {
2454 int version;
2455 if (!GetRtpVersion(packet->data(), packet->length(), &version)) {
2456 return false;
2457 }
2458
2459 return version == 2;
2460}
2461
2462bool DataChannel::WantsPacket(bool rtcp, talk_base::Buffer* packet) {
2463 if (data_channel_type_ == DCT_SCTP) {
2464 // TODO(pthatcher): Do this in a more robust way by checking for
2465 // SCTP or DTLS.
2466 return !IsRtpPacket(packet);
2467 } else if (data_channel_type_ == DCT_RTP) {
2468 return BaseChannel::WantsPacket(rtcp, packet);
2469 }
2470 return false;
2471}
2472
2473// Sets the maximum bandwidth. Anything over this will be dropped.
2474bool DataChannel::SetMaxSendBandwidth_w(int max_bps) {
2475 LOG(LS_INFO) << "DataChannel: Setting max bandwidth to " << max_bps;
2476 return media_channel()->SetSendBandwidth(false, max_bps);
2477}
2478
2479bool DataChannel::SetDataChannelType(DataChannelType new_data_channel_type) {
2480 // It hasn't been set before, so set it now.
2481 if (data_channel_type_ == DCT_NONE) {
2482 data_channel_type_ = new_data_channel_type;
2483 return true;
2484 }
2485
2486 // It's been set before, but doesn't match. That's bad.
2487 if (data_channel_type_ != new_data_channel_type) {
2488 LOG(LS_WARNING) << "Data channel type mismatch."
2489 << " Expected " << data_channel_type_
2490 << " Got " << new_data_channel_type;
2491 return false;
2492 }
2493
2494 // It's hasn't changed. Nothing to do.
2495 return true;
2496}
2497
2498bool DataChannel::SetDataChannelTypeFromContent(
2499 const DataContentDescription* content) {
2500 bool is_sctp = ((content->protocol() == kMediaProtocolSctp) ||
2501 (content->protocol() == kMediaProtocolDtlsSctp));
2502 DataChannelType data_channel_type = is_sctp ? DCT_SCTP : DCT_RTP;
2503 return SetDataChannelType(data_channel_type);
2504}
2505
2506bool DataChannel::SetLocalContent_w(const MediaContentDescription* content,
2507 ContentAction action) {
2508 ASSERT(worker_thread() == talk_base::Thread::Current());
2509 LOG(LS_INFO) << "Setting local data description";
2510
2511 const DataContentDescription* data =
2512 static_cast<const DataContentDescription*>(content);
2513 ASSERT(data != NULL);
2514 if (!data) return false;
2515
2516 bool ret = false;
2517 if (!SetDataChannelTypeFromContent(data)) {
2518 return false;
2519 }
2520
2521 if (data_channel_type_ == DCT_SCTP) {
2522 // SCTP data channels don't need the rest of the stuff.
2523 ret = UpdateLocalStreams_w(data->streams(), action);
2524 if (ret) {
2525 set_local_content_direction(content->direction());
2526 }
2527 } else {
2528 ret = SetBaseLocalContent_w(content, action);
2529
2530 if (action != CA_UPDATE || data->has_codecs()) {
2531 ret &= media_channel()->SetRecvCodecs(data->codecs());
2532 }
2533 }
2534
2535 // If everything worked, see if we can start receiving.
2536 if (ret) {
2537 ChangeState();
2538 } else {
2539 LOG(LS_WARNING) << "Failed to set local data description";
2540 }
2541 return ret;
2542}
2543
2544bool DataChannel::SetRemoteContent_w(const MediaContentDescription* content,
2545 ContentAction action) {
2546 ASSERT(worker_thread() == talk_base::Thread::Current());
2547
2548 const DataContentDescription* data =
2549 static_cast<const DataContentDescription*>(content);
2550 ASSERT(data != NULL);
2551 if (!data) return false;
2552
2553 bool ret = true;
2554 if (!SetDataChannelTypeFromContent(data)) {
2555 return false;
2556 }
2557
2558 if (data_channel_type_ == DCT_SCTP) {
2559 LOG(LS_INFO) << "Setting SCTP remote data description";
2560 // SCTP data channels don't need the rest of the stuff.
2561 ret = UpdateRemoteStreams_w(content->streams(), action);
2562 if (ret) {
2563 set_remote_content_direction(content->direction());
2564 }
2565 } else {
2566 // If the remote data doesn't have codecs and isn't an update, it
2567 // must be empty, so ignore it.
2568 if (action != CA_UPDATE && !data->has_codecs()) {
2569 return true;
2570 }
2571 LOG(LS_INFO) << "Setting remote data description";
2572
2573 // Set remote video codecs (what the other side wants to receive).
2574 if (action != CA_UPDATE || data->has_codecs()) {
2575 ret &= media_channel()->SetSendCodecs(data->codecs());
2576 }
2577
2578 if (ret) {
2579 ret &= SetBaseRemoteContent_w(content, action);
2580 }
2581
2582 if (action != CA_UPDATE) {
2583 int bandwidth_bps = data->bandwidth();
2584 bool auto_bandwidth = (bandwidth_bps == kAutoBandwidth);
2585 ret &= media_channel()->SetSendBandwidth(auto_bandwidth, bandwidth_bps);
2586 }
2587 }
2588
2589 // If everything worked, see if we can start sending.
2590 if (ret) {
2591 ChangeState();
2592 } else {
2593 LOG(LS_WARNING) << "Failed to set remote data description";
2594 }
2595 return ret;
2596}
2597
2598void DataChannel::ChangeState() {
2599 // Render incoming data if we're the active call, and we have the local
2600 // content. We receive data on the default channel and multiplexed streams.
2601 bool recv = IsReadyToReceive();
2602 if (!media_channel()->SetReceive(recv)) {
2603 LOG(LS_ERROR) << "Failed to SetReceive on data channel";
2604 }
2605
2606 // Send outgoing data if we're the active call, we have the remote content,
2607 // and we have had some form of connectivity.
2608 bool send = IsReadyToSend();
2609 if (!media_channel()->SetSend(send)) {
2610 LOG(LS_ERROR) << "Failed to SetSend on data channel";
2611 }
2612
2613 // Post to trigger SignalReadyToSendData.
2614 signaling_thread()->Post(this, MSG_READYTOSENDDATA,
wu@webrtc.orgd64719d2013-08-01 00:00:07 +00002615 new DataChannelReadyToSendMessageData(send));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002616
2617 LOG(LS_INFO) << "Changing data state, recv=" << recv << " send=" << send;
2618}
2619
2620void DataChannel::OnMessage(talk_base::Message *pmsg) {
2621 switch (pmsg->message_id) {
2622 case MSG_READYTOSENDDATA: {
wu@webrtc.orgd64719d2013-08-01 00:00:07 +00002623 DataChannelReadyToSendMessageData* data =
2624 static_cast<DataChannelReadyToSendMessageData*>(pmsg->pdata);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002625 SignalReadyToSendData(data->data());
2626 delete data;
2627 break;
2628 }
2629 case MSG_SENDDATA: {
2630 SendDataMessageData* msg =
2631 static_cast<SendDataMessageData*>(pmsg->pdata);
2632 msg->succeeded = media_channel()->SendData(
2633 msg->params, *(msg->payload), msg->result);
2634 break;
2635 }
2636 case MSG_DATARECEIVED: {
2637 DataReceivedMessageData* data =
2638 static_cast<DataReceivedMessageData*>(pmsg->pdata);
2639 SignalDataReceived(this, data->params, data->payload);
2640 delete data;
2641 break;
2642 }
2643 case MSG_CHANNEL_ERROR: {
2644 const DataChannelErrorMessageData* data =
2645 static_cast<DataChannelErrorMessageData*>(pmsg->pdata);
2646 SignalMediaError(this, data->ssrc, data->error);
2647 delete data;
2648 break;
2649 }
2650 default:
2651 BaseChannel::OnMessage(pmsg);
2652 break;
2653 }
2654}
2655
2656void DataChannel::OnConnectionMonitorUpdate(
2657 SocketMonitor* monitor, const std::vector<ConnectionInfo>& infos) {
2658 SignalConnectionMonitor(this, infos);
2659}
2660
2661void DataChannel::StartMediaMonitor(int cms) {
2662 media_monitor_.reset(new DataMediaMonitor(media_channel(), worker_thread(),
2663 talk_base::Thread::Current()));
2664 media_monitor_->SignalUpdate.connect(
2665 this, &DataChannel::OnMediaMonitorUpdate);
2666 media_monitor_->Start(cms);
2667}
2668
2669void DataChannel::StopMediaMonitor() {
2670 if (media_monitor_) {
2671 media_monitor_->Stop();
2672 media_monitor_->SignalUpdate.disconnect(this);
2673 media_monitor_.reset();
2674 }
2675}
2676
2677void DataChannel::OnMediaMonitorUpdate(
2678 DataMediaChannel* media_channel, const DataMediaInfo& info) {
2679 ASSERT(media_channel == this->media_channel());
2680 SignalMediaMonitor(this, info);
2681}
2682
2683void DataChannel::OnDataReceived(
2684 const ReceiveDataParams& params, const char* data, size_t len) {
2685 DataReceivedMessageData* msg = new DataReceivedMessageData(
2686 params, data, len);
2687 signaling_thread()->Post(this, MSG_DATARECEIVED, msg);
2688}
2689
2690void DataChannel::OnDataChannelError(
2691 uint32 ssrc, DataMediaChannel::Error err) {
2692 DataChannelErrorMessageData* data = new DataChannelErrorMessageData(
2693 ssrc, err);
2694 signaling_thread()->Post(this, MSG_CHANNEL_ERROR, data);
2695}
2696
wu@webrtc.orgd64719d2013-08-01 00:00:07 +00002697void DataChannel::OnDataChannelReadyToSend(bool writable) {
2698 // This is usded for congestion control to indicate that the stream is ready
2699 // to send by the MediaChannel, as opposed to OnReadyToSend, which indicates
2700 // that the transport channel is ready.
2701 signaling_thread()->Post(this, MSG_READYTOSENDDATA,
2702 new DataChannelReadyToSendMessageData(writable));
2703}
2704
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002705void DataChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode,
2706 SrtpFilter::Error error) {
2707 switch (error) {
2708 case SrtpFilter::ERROR_FAIL:
2709 OnDataChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
2710 DataMediaChannel::ERROR_SEND_SRTP_ERROR :
2711 DataMediaChannel::ERROR_RECV_SRTP_ERROR);
2712 break;
2713 case SrtpFilter::ERROR_AUTH:
2714 OnDataChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
2715 DataMediaChannel::ERROR_SEND_SRTP_AUTH_FAILED :
2716 DataMediaChannel::ERROR_RECV_SRTP_AUTH_FAILED);
2717 break;
2718 case SrtpFilter::ERROR_REPLAY:
2719 // Only receving channel should have this error.
2720 ASSERT(mode == SrtpFilter::UNPROTECT);
2721 OnDataChannelError(ssrc, DataMediaChannel::ERROR_RECV_SRTP_REPLAY);
2722 break;
2723 default:
2724 break;
2725 }
2726}
2727
2728void DataChannel::GetSrtpCiphers(std::vector<std::string>* ciphers) const {
2729 GetSupportedDataCryptoSuites(ciphers);
2730}
2731
2732bool DataChannel::ShouldSetupDtlsSrtp() const {
2733 return (data_channel_type_ == DCT_RTP);
2734}
2735
2736} // namespace cricket