forked from HackWebRTC/webrtc
-
Notifications
You must be signed in to change notification settings - Fork 0
/
Copy pathchannel.cc
1373 lines (1200 loc) · 48.8 KB
/
channel.cc
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000
/*
* Copyright 2004 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/channel.h"
#include <iterator>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "api/call/audio_sink.h"
#include "api/transport/media/media_transport_config.h"
#include "media/base/media_constants.h"
#include "media/base/rtp_utils.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "p2p/base/packet_transport_internal.h"
#include "pc/channel_manager.h"
#include "pc/rtp_media_utils.h"
#include "rtc_base/bind.h"
#include "rtc_base/byte_order.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/dscp.h"
#include "rtc_base/logging.h"
#include "rtc_base/network_route.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/trace_event.h"
namespace cricket {
using rtc::Bind;
using rtc::UniqueRandomIdGenerator;
using webrtc::SdpType;
namespace {
struct SendPacketMessageData : public rtc::MessageData {
rtc::CopyOnWriteBuffer packet;
rtc::PacketOptions options;
};
// Finds a stream based on target's Primary SSRC or RIDs.
// This struct is used in BaseChannel::UpdateLocalStreams_w.
struct StreamFinder {
explicit StreamFinder(const StreamParams* target) : target_(target) {
RTC_DCHECK(target);
}
bool operator()(const StreamParams& sp) const {
if (target_->has_ssrcs() && sp.has_ssrcs()) {
return sp.has_ssrc(target_->first_ssrc());
}
if (!target_->has_rids() && !sp.has_rids()) {
return false;
}
const std::vector<RidDescription>& target_rids = target_->rids();
const std::vector<RidDescription>& source_rids = sp.rids();
if (source_rids.size() != target_rids.size()) {
return false;
}
// Check that all RIDs match.
return std::equal(source_rids.begin(), source_rids.end(),
target_rids.begin(),
[](const RidDescription& lhs, const RidDescription& rhs) {
return lhs.rid == rhs.rid;
});
}
const StreamParams* target_;
};
} // namespace
enum {
MSG_SEND_RTP_PACKET = 1,
MSG_SEND_RTCP_PACKET,
MSG_READYTOSENDDATA,
MSG_DATARECEIVED,
MSG_FIRSTPACKETRECEIVED,
};
static void SafeSetError(const std::string& message, std::string* error_desc) {
if (error_desc) {
*error_desc = message;
}
}
template <class Codec>
void RtpParametersFromMediaDescription(
const MediaContentDescriptionImpl<Codec>* desc,
const RtpHeaderExtensions& extensions,
bool is_stream_active,
RtpParameters<Codec>* params) {
params->is_stream_active = is_stream_active;
// TODO(pthatcher): Remove this once we're sure no one will give us
// a description without codecs. Currently the ORTC implementation is relying
// on this.
if (desc->has_codecs()) {
params->codecs = desc->codecs();
}
// TODO(pthatcher): See if we really need
// rtp_header_extensions_set() and remove it if we don't.
if (desc->rtp_header_extensions_set()) {
params->extensions = extensions;
}
params->rtcp.reduced_size = desc->rtcp_reduced_size();
params->rtcp.remote_estimate = desc->remote_estimate();
}
template <class Codec>
void RtpSendParametersFromMediaDescription(
const MediaContentDescriptionImpl<Codec>* desc,
const RtpHeaderExtensions& extensions,
bool is_stream_active,
RtpSendParameters<Codec>* send_params) {
RtpParametersFromMediaDescription(desc, extensions, is_stream_active,
send_params);
send_params->max_bandwidth_bps = desc->bandwidth();
send_params->extmap_allow_mixed = desc->extmap_allow_mixed();
}
BaseChannel::BaseChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<MediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
UniqueRandomIdGenerator* ssrc_generator)
: worker_thread_(worker_thread),
network_thread_(network_thread),
signaling_thread_(signaling_thread),
content_name_(content_name),
srtp_required_(srtp_required),
crypto_options_(crypto_options),
media_channel_(std::move(media_channel)),
ssrc_generator_(ssrc_generator) {
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_DCHECK(ssrc_generator_);
demuxer_criteria_.mid = content_name;
RTC_LOG(LS_INFO) << "Created channel for " << content_name;
}
BaseChannel::~BaseChannel() {
TRACE_EVENT0("webrtc", "BaseChannel::~BaseChannel");
RTC_DCHECK_RUN_ON(worker_thread_);
// Eats any outstanding messages or packets.
worker_thread_->Clear(&invoker_);
worker_thread_->Clear(this);
// We must destroy the media channel before the transport channel, otherwise
// the media channel may try to send on the dead transport channel. NULLing
// is not an effective strategy since the sends will come on another thread.
media_channel_.reset();
RTC_LOG(LS_INFO) << "Destroyed channel: " << content_name_;
}
bool BaseChannel::ConnectToRtpTransport() {
RTC_DCHECK(rtp_transport_);
if (!RegisterRtpDemuxerSink()) {
return false;
}
rtp_transport_->SignalReadyToSend.connect(
this, &BaseChannel::OnTransportReadyToSend);
rtp_transport_->SignalNetworkRouteChanged.connect(
this, &BaseChannel::OnNetworkRouteChanged);
rtp_transport_->SignalWritableState.connect(this,
&BaseChannel::OnWritableState);
rtp_transport_->SignalSentPacket.connect(this,
&BaseChannel::SignalSentPacket_n);
return true;
}
void BaseChannel::DisconnectFromRtpTransport() {
RTC_DCHECK(rtp_transport_);
rtp_transport_->UnregisterRtpDemuxerSink(this);
rtp_transport_->SignalReadyToSend.disconnect(this);
rtp_transport_->SignalNetworkRouteChanged.disconnect(this);
rtp_transport_->SignalWritableState.disconnect(this);
rtp_transport_->SignalSentPacket.disconnect(this);
}
void BaseChannel::Init_w(
webrtc::RtpTransportInternal* rtp_transport,
const webrtc::MediaTransportConfig& media_transport_config) {
RTC_DCHECK_RUN_ON(worker_thread_);
media_transport_config_ = media_transport_config;
network_thread_->Invoke<void>(
RTC_FROM_HERE, [this, rtp_transport] { SetRtpTransport(rtp_transport); });
// Both RTP and RTCP channels should be set, we can call SetInterface on
// the media channel and it can set network options.
media_channel_->SetInterface(this, media_transport_config);
}
void BaseChannel::Deinit() {
RTC_DCHECK(worker_thread_->IsCurrent());
media_channel_->SetInterface(/*iface=*/nullptr,
webrtc::MediaTransportConfig());
// Packets arrive on the network thread, processing packets calls virtual
// functions, so need to stop this process in Deinit that is called in
// derived classes destructor.
network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
FlushRtcpMessages_n();
if (rtp_transport_) {
DisconnectFromRtpTransport();
}
// Clear pending read packets/messages.
network_thread_->Clear(&invoker_);
network_thread_->Clear(this);
});
}
bool BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) {
if (rtp_transport == rtp_transport_) {
return true;
}
if (!network_thread_->IsCurrent()) {
return network_thread_->Invoke<bool>(RTC_FROM_HERE, [this, rtp_transport] {
return SetRtpTransport(rtp_transport);
});
}
if (rtp_transport_) {
DisconnectFromRtpTransport();
}
rtp_transport_ = rtp_transport;
if (rtp_transport_) {
transport_name_ = rtp_transport_->transport_name();
if (!ConnectToRtpTransport()) {
RTC_LOG(LS_ERROR) << "Failed to connect to the new RtpTransport.";
return false;
}
OnTransportReadyToSend(rtp_transport_->IsReadyToSend());
UpdateWritableState_n();
// Set the cached socket options.
for (const auto& pair : socket_options_) {
rtp_transport_->SetRtpOption(pair.first, pair.second);
}
if (!rtp_transport_->rtcp_mux_enabled()) {
for (const auto& pair : rtcp_socket_options_) {
rtp_transport_->SetRtcpOption(pair.first, pair.second);
}
}
}
return true;
}
bool BaseChannel::Enable(bool enable) {
worker_thread_->Invoke<void>(
RTC_FROM_HERE,
Bind(enable ? &BaseChannel::EnableMedia_w : &BaseChannel::DisableMedia_w,
this));
return true;
}
bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent");
return InvokeOnWorker<bool>(
RTC_FROM_HERE,
Bind(&BaseChannel::SetLocalContent_w, this, content, type, error_desc));
}
bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent");
return InvokeOnWorker<bool>(
RTC_FROM_HERE,
Bind(&BaseChannel::SetRemoteContent_w, this, content, type, error_desc));
}
bool BaseChannel::IsReadyToReceiveMedia_w() const {
// Receive data if we are enabled and have local content,
return enabled() &&
webrtc::RtpTransceiverDirectionHasRecv(local_content_direction_);
}
bool BaseChannel::IsReadyToSendMedia_w() const {
// Need to access some state updated on the network thread.
return network_thread_->Invoke<bool>(
RTC_FROM_HERE, Bind(&BaseChannel::IsReadyToSendMedia_n, this));
}
bool BaseChannel::IsReadyToSendMedia_n() const {
// Send outgoing data if we are enabled, have local and remote content,
// and we have had some form of connectivity.
return enabled() &&
webrtc::RtpTransceiverDirectionHasRecv(remote_content_direction_) &&
webrtc::RtpTransceiverDirectionHasSend(local_content_direction_) &&
was_ever_writable();
}
bool BaseChannel::SendPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) {
return SendPacket(false, packet, options);
}
bool BaseChannel::SendRtcp(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) {
return SendPacket(true, packet, options);
}
int BaseChannel::SetOption(SocketType type,
rtc::Socket::Option opt,
int value) {
return network_thread_->Invoke<int>(
RTC_FROM_HERE, Bind(&BaseChannel::SetOption_n, this, type, opt, value));
}
int BaseChannel::SetOption_n(SocketType type,
rtc::Socket::Option opt,
int value) {
RTC_DCHECK(network_thread_->IsCurrent());
RTC_DCHECK(rtp_transport_);
switch (type) {
case ST_RTP:
socket_options_.push_back(
std::pair<rtc::Socket::Option, int>(opt, value));
return rtp_transport_->SetRtpOption(opt, value);
case ST_RTCP:
rtcp_socket_options_.push_back(
std::pair<rtc::Socket::Option, int>(opt, value));
return rtp_transport_->SetRtcpOption(opt, value);
}
return -1;
}
void BaseChannel::OnWritableState(bool writable) {
RTC_DCHECK(network_thread_->IsCurrent());
if (writable) {
ChannelWritable_n();
} else {
ChannelNotWritable_n();
}
}
void BaseChannel::OnNetworkRouteChanged(
absl::optional<rtc::NetworkRoute> network_route) {
RTC_LOG(LS_INFO) << "Network route was changed.";
RTC_DCHECK(network_thread_->IsCurrent());
rtc::NetworkRoute new_route;
if (network_route) {
new_route = *(network_route);
}
// Note: When the RTCP-muxing is not enabled, RTCP transport and RTP transport
// use the same transport name and MediaChannel::OnNetworkRouteChanged cannot
// work correctly. Intentionally leave it broken to simplify the code and
// encourage the users to stop using non-muxing RTCP.
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_, [=] {
media_channel_->OnNetworkRouteChanged(transport_name_, new_route);
});
}
void BaseChannel::OnTransportReadyToSend(bool ready) {
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_,
[=] { media_channel_->OnReadyToSend(ready); });
}
bool BaseChannel::SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) {
// Until all the code is migrated to use RtpPacketType instead of bool.
RtpPacketType packet_type = rtcp ? RtpPacketType::kRtcp : RtpPacketType::kRtp;
// SendPacket gets called from MediaEngine, on a pacer or an encoder thread.
// If the thread is not our network thread, we will post to our network
// so that the real work happens on our network. This avoids us having to
// synchronize access to all the pieces of the send path, including
// SRTP and the inner workings of the transport channels.
// The only downside is that we can't return a proper failure code if
// needed. Since UDP is unreliable anyway, this should be a non-issue.
if (!network_thread_->IsCurrent()) {
// Avoid a copy by transferring the ownership of the packet data.
int message_id = rtcp ? MSG_SEND_RTCP_PACKET : MSG_SEND_RTP_PACKET;
SendPacketMessageData* data = new SendPacketMessageData;
data->packet = std::move(*packet);
data->options = options;
network_thread_->Post(RTC_FROM_HERE, this, message_id, data);
return true;
}
TRACE_EVENT0("webrtc", "BaseChannel::SendPacket");
// Now that we are on the correct thread, ensure we have a place to send this
// packet before doing anything. (We might get RTCP packets that we don't
// intend to send.) If we've negotiated RTCP mux, send RTCP over the RTP
// transport.
if (!rtp_transport_ || !rtp_transport_->IsWritable(rtcp)) {
return false;
}
// Protect ourselves against crazy data.
if (!IsValidRtpPacketSize(packet_type, packet->size())) {
RTC_LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " "
<< RtpPacketTypeToString(packet_type)
<< " packet: wrong size=" << packet->size();
return false;
}
if (!srtp_active()) {
if (srtp_required_) {
// The audio/video engines may attempt to send RTCP packets as soon as the
// streams are created, so don't treat this as an error for RTCP.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6809
if (rtcp) {
return false;
}
// However, there shouldn't be any RTP packets sent before SRTP is set up
// (and SetSend(true) is called).
RTC_LOG(LS_ERROR)
<< "Can't send outgoing RTP packet when SRTP is inactive"
" and crypto is required";
RTC_NOTREACHED();
return false;
}
std::string packet_type = rtcp ? "RTCP" : "RTP";
RTC_LOG(LS_WARNING) << "Sending an " << packet_type
<< " packet without encryption.";
}
// Bon voyage.
return rtcp ? rtp_transport_->SendRtcpPacket(packet, options, PF_SRTP_BYPASS)
: rtp_transport_->SendRtpPacket(packet, options, PF_SRTP_BYPASS);
}
void BaseChannel::OnRtpPacket(const webrtc::RtpPacketReceived& parsed_packet) {
// Take packet time from the |parsed_packet|.
// RtpPacketReceived.arrival_time_ms = (timestamp_us + 500) / 1000;
int64_t packet_time_us = -1;
if (parsed_packet.arrival_time_ms() > 0) {
packet_time_us = parsed_packet.arrival_time_ms() * 1000;
}
if (!has_received_packet_) {
has_received_packet_ = true;
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FIRSTPACKETRECEIVED);
}
if (!srtp_active() && srtp_required_) {
// Our session description indicates that SRTP is required, but we got a
// packet before our SRTP filter is active. This means either that
// a) we got SRTP packets before we received the SDES keys, in which case
// we can't decrypt it anyway, or
// b) we got SRTP packets before DTLS completed on both the RTP and RTCP
// transports, so we haven't yet extracted keys, even if DTLS did
// complete on the transport that the packets are being sent on. It's
// really good practice to wait for both RTP and RTCP to be good to go
// before sending media, to prevent weird failure modes, so it's fine
// for us to just eat packets here. This is all sidestepped if RTCP mux
// is used anyway.
RTC_LOG(LS_WARNING) << "Can't process incoming RTP packet when "
"SRTP is inactive and crypto is required";
return;
}
auto packet_buffer = parsed_packet.Buffer();
invoker_.AsyncInvoke<void>(
RTC_FROM_HERE, worker_thread_, [this, packet_buffer, packet_time_us] {
RTC_DCHECK(worker_thread_->IsCurrent());
media_channel_->OnPacketReceived(packet_buffer, packet_time_us);
});
}
void BaseChannel::UpdateRtpHeaderExtensionMap(
const RtpHeaderExtensions& header_extensions) {
RTC_DCHECK(rtp_transport_);
// Update the header extension map on network thread in case there is data
// race.
// TODO(zhihuang): Add an rtc::ThreadChecker make sure to RtpTransport won't
// be accessed from different threads.
//
// NOTE: This doesn't take the BUNDLE case in account meaning the RTP header
// extension maps are not merged when BUNDLE is enabled. This is fine because
// the ID for MID should be consistent among all the RTP transports.
network_thread_->Invoke<void>(RTC_FROM_HERE, [this, &header_extensions] {
rtp_transport_->UpdateRtpHeaderExtensionMap(header_extensions);
});
}
bool BaseChannel::RegisterRtpDemuxerSink() {
RTC_DCHECK(rtp_transport_);
return network_thread_->Invoke<bool>(RTC_FROM_HERE, [this] {
return rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria_, this);
});
}
void BaseChannel::EnableMedia_w() {
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
if (enabled_)
return;
RTC_LOG(LS_INFO) << "Channel enabled";
enabled_ = true;
UpdateMediaSendRecvState_w();
}
void BaseChannel::DisableMedia_w() {
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
if (!enabled_)
return;
RTC_LOG(LS_INFO) << "Channel disabled";
enabled_ = false;
UpdateMediaSendRecvState_w();
}
void BaseChannel::UpdateWritableState_n() {
if (rtp_transport_->IsWritable(/*rtcp=*/true) &&
rtp_transport_->IsWritable(/*rtcp=*/false)) {
ChannelWritable_n();
} else {
ChannelNotWritable_n();
}
}
void BaseChannel::ChannelWritable_n() {
RTC_DCHECK(network_thread_->IsCurrent());
if (writable_) {
return;
}
RTC_LOG(LS_INFO) << "Channel writable (" << content_name_ << ")"
<< (was_ever_writable_ ? "" : " for the first time");
was_ever_writable_ = true;
writable_ = true;
UpdateMediaSendRecvState();
}
void BaseChannel::ChannelNotWritable_n() {
RTC_DCHECK(network_thread_->IsCurrent());
if (!writable_)
return;
RTC_LOG(LS_INFO) << "Channel not writable (" << content_name_ << ")";
writable_ = false;
UpdateMediaSendRecvState();
}
bool BaseChannel::AddRecvStream_w(const StreamParams& sp) {
RTC_DCHECK(worker_thread() == rtc::Thread::Current());
return media_channel()->AddRecvStream(sp);
}
bool BaseChannel::RemoveRecvStream_w(uint32_t ssrc) {
RTC_DCHECK(worker_thread() == rtc::Thread::Current());
return media_channel()->RemoveRecvStream(ssrc);
}
void BaseChannel::ResetUnsignaledRecvStream_w() {
RTC_DCHECK(worker_thread() == rtc::Thread::Current());
media_channel()->ResetUnsignaledRecvStream();
}
bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
SdpType type,
std::string* error_desc) {
// In the case of RIDs (where SSRCs are not negotiated), this method will
// generate an SSRC for each layer in StreamParams. That representation will
// be stored internally in |local_streams_|.
// In subsequent offers, the same stream can appear in |streams| again
// (without the SSRCs), so it should be looked up using RIDs (if available)
// and then by primary SSRC.
// In both scenarios, it is safe to assume that the media channel will be
// created with a StreamParams object with SSRCs. However, it is not safe to
// assume that |local_streams_| will always have SSRCs as there are scenarios
// in which niether SSRCs or RIDs are negotiated.
// Check for streams that have been removed.
bool ret = true;
for (const StreamParams& old_stream : local_streams_) {
if (!old_stream.has_ssrcs() ||
GetStream(streams, StreamFinder(&old_stream))) {
continue;
}
if (!media_channel()->RemoveSendStream(old_stream.first_ssrc())) {
rtc::StringBuilder desc;
desc << "Failed to remove send stream with ssrc "
<< old_stream.first_ssrc() << ".";
SafeSetError(desc.str(), error_desc);
ret = false;
}
}
// Check for new streams.
std::vector<StreamParams> all_streams;
for (const StreamParams& stream : streams) {
StreamParams* existing = GetStream(local_streams_, StreamFinder(&stream));
if (existing) {
// Parameters cannot change for an existing stream.
all_streams.push_back(*existing);
continue;
}
all_streams.push_back(stream);
StreamParams& new_stream = all_streams.back();
if (!new_stream.has_ssrcs() && !new_stream.has_rids()) {
continue;
}
RTC_DCHECK(new_stream.has_ssrcs() || new_stream.has_rids());
if (new_stream.has_ssrcs() && new_stream.has_rids()) {
rtc::StringBuilder desc;
desc << "Failed to add send stream: " << new_stream.first_ssrc()
<< ". Stream has both SSRCs and RIDs.";
SafeSetError(desc.str(), error_desc);
ret = false;
continue;
}
// At this point we use the legacy simulcast group in StreamParams to
// indicate that we want multiple layers to the media channel.
if (!new_stream.has_ssrcs()) {
// TODO(bugs.webrtc.org/10250): Indicate if flex is desired here.
new_stream.GenerateSsrcs(new_stream.rids().size(), /* rtx = */ true,
/* flex_fec = */ false, ssrc_generator_);
}
if (media_channel()->AddSendStream(new_stream)) {
RTC_LOG(LS_INFO) << "Add send stream ssrc: " << new_stream.ssrcs[0];
} else {
rtc::StringBuilder desc;
desc << "Failed to add send stream ssrc: " << new_stream.first_ssrc();
SafeSetError(desc.str(), error_desc);
ret = false;
}
}
local_streams_ = all_streams;
return ret;
}
bool BaseChannel::UpdateRemoteStreams_w(
const std::vector<StreamParams>& streams,
SdpType type,
std::string* error_desc) {
// Check for streams that have been removed.
bool ret = true;
for (const StreamParams& old_stream : remote_streams_) {
// If we no longer have an unsignaled stream, we would like to remove
// the unsignaled stream params that are cached.
if (!old_stream.has_ssrcs() && !HasStreamWithNoSsrcs(streams)) {
ResetUnsignaledRecvStream_w();
RTC_LOG(LS_INFO) << "Reset unsignaled remote stream.";
} else if (old_stream.has_ssrcs() &&
!GetStreamBySsrc(streams, old_stream.first_ssrc())) {
if (RemoveRecvStream_w(old_stream.first_ssrc())) {
RTC_LOG(LS_INFO) << "Remove remote ssrc: " << old_stream.first_ssrc();
} else {
rtc::StringBuilder desc;
desc << "Failed to remove remote stream with ssrc "
<< old_stream.first_ssrc() << ".";
SafeSetError(desc.str(), error_desc);
ret = false;
}
}
}
demuxer_criteria_.ssrcs.clear();
// Check for new streams.
for (const StreamParams& new_stream : streams) {
// We allow a StreamParams with an empty list of SSRCs, in which case the
// MediaChannel will cache the parameters and use them for any unsignaled
// stream received later.
if ((!new_stream.has_ssrcs() && !HasStreamWithNoSsrcs(remote_streams_)) ||
!GetStreamBySsrc(remote_streams_, new_stream.first_ssrc())) {
if (AddRecvStream_w(new_stream)) {
RTC_LOG(LS_INFO) << "Add remote ssrc: "
<< (new_stream.has_ssrcs()
? std::to_string(new_stream.first_ssrc())
: "unsignaled");
} else {
rtc::StringBuilder desc;
desc << "Failed to add remote stream ssrc: "
<< (new_stream.has_ssrcs()
? std::to_string(new_stream.first_ssrc())
: "unsignaled");
SafeSetError(desc.str(), error_desc);
ret = false;
}
}
// Update the receiving SSRCs.
demuxer_criteria_.ssrcs.insert(new_stream.ssrcs.begin(),
new_stream.ssrcs.end());
}
// Re-register the sink to update the receiving ssrcs.
RegisterRtpDemuxerSink();
remote_streams_ = streams;
return ret;
}
RtpHeaderExtensions BaseChannel::GetFilteredRtpHeaderExtensions(
const RtpHeaderExtensions& extensions) {
RTC_DCHECK(rtp_transport_);
if (crypto_options_.srtp.enable_encrypted_rtp_header_extensions) {
RtpHeaderExtensions filtered;
absl::c_copy_if(extensions, std::back_inserter(filtered),
[](const webrtc::RtpExtension& extension) {
return !extension.encrypt;
});
return filtered;
}
return webrtc::RtpExtension::FilterDuplicateNonEncrypted(extensions);
}
void BaseChannel::OnMessage(rtc::Message* pmsg) {
TRACE_EVENT0("webrtc", "BaseChannel::OnMessage");
switch (pmsg->message_id) {
case MSG_SEND_RTP_PACKET:
case MSG_SEND_RTCP_PACKET: {
RTC_DCHECK(network_thread_->IsCurrent());
SendPacketMessageData* data =
static_cast<SendPacketMessageData*>(pmsg->pdata);
bool rtcp = pmsg->message_id == MSG_SEND_RTCP_PACKET;
SendPacket(rtcp, &data->packet, data->options);
delete data;
break;
}
case MSG_FIRSTPACKETRECEIVED: {
SignalFirstPacketReceived_(this);
break;
}
}
}
void BaseChannel::AddHandledPayloadType(int payload_type) {
demuxer_criteria_.payload_types.insert(static_cast<uint8_t>(payload_type));
}
void BaseChannel::ClearHandledPayloadTypes() {
demuxer_criteria_.payload_types.clear();
}
void BaseChannel::FlushRtcpMessages_n() {
// Flush all remaining RTCP messages. This should only be called in
// destructor.
RTC_DCHECK(network_thread_->IsCurrent());
rtc::MessageList rtcp_messages;
network_thread_->Clear(this, MSG_SEND_RTCP_PACKET, &rtcp_messages);
for (const auto& message : rtcp_messages) {
network_thread_->Send(RTC_FROM_HERE, this, MSG_SEND_RTCP_PACKET,
message.pdata);
}
}
void BaseChannel::SignalSentPacket_n(const rtc::SentPacket& sent_packet) {
RTC_DCHECK(network_thread_->IsCurrent());
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_,
[this, sent_packet] {
RTC_DCHECK(worker_thread_->IsCurrent());
SignalSentPacket(sent_packet);
});
}
VoiceChannel::VoiceChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VoiceMediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
UniqueRandomIdGenerator* ssrc_generator)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
std::move(media_channel),
content_name,
srtp_required,
crypto_options,
ssrc_generator) {}
VoiceChannel::~VoiceChannel() {
TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel");
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
Deinit();
}
void BaseChannel::UpdateMediaSendRecvState() {
RTC_DCHECK(network_thread_->IsCurrent());
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_,
[this] { UpdateMediaSendRecvState_w(); });
}
void VoiceChannel::Init_w(
webrtc::RtpTransportInternal* rtp_transport,
const webrtc::MediaTransportConfig& media_transport_config) {
BaseChannel::Init_w(rtp_transport, media_transport_config);
}
void VoiceChannel::UpdateMediaSendRecvState_w() {
// Render incoming data if we're the active call, and we have the local
// content. We receive data on the default channel and multiplexed streams.
bool recv = IsReadyToReceiveMedia_w();
media_channel()->SetPlayout(recv);
// Send outgoing data if we're the active call, we have the remote content,
// and we have had some form of connectivity.
bool send = IsReadyToSendMedia_w();
media_channel()->SetSend(send);
RTC_LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send;
}
bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VoiceChannel::SetLocalContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting local voice description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find audio content in local description.", error_desc);
return false;
}
const AudioContentDescription* audio = content->as_audio();
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
UpdateRtpHeaderExtensionMap(rtp_header_extensions);
media_channel()->SetExtmapAllowMixed(audio->extmap_allow_mixed());
AudioRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(
audio, rtp_header_extensions,
webrtc::RtpTransceiverDirectionHasRecv(audio->direction()), &recv_params);
if (!media_channel()->SetRecvParameters(recv_params)) {
SafeSetError("Failed to set local audio description recv parameters.",
error_desc);
return false;
}
if (webrtc::RtpTransceiverDirectionHasRecv(audio->direction())) {
for (const AudioCodec& codec : audio->codecs()) {
AddHandledPayloadType(codec.id);
}
// Need to re-register the sink to update the handled payload.
if (!RegisterRtpDemuxerSink()) {
RTC_LOG(LS_ERROR) << "Failed to set up audio demuxing.";
return false;
}
}
last_recv_params_ = recv_params;
// TODO(pthatcher): Move local streams into AudioSendParameters, and
// only give it to the media channel once we have a remote
// description too (without a remote description, we won't be able
// to send them anyway).
if (!UpdateLocalStreams_w(audio->streams(), type, error_desc)) {
SafeSetError("Failed to set local audio description streams.", error_desc);
return false;
}
set_local_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VoiceChannel::SetRemoteContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting remote voice description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find audio content in remote description.", error_desc);
return false;
}
const AudioContentDescription* audio = content->as_audio();
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
AudioSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription(
audio, rtp_header_extensions,
webrtc::RtpTransceiverDirectionHasRecv(audio->direction()), &send_params);
send_params.mid = content_name();
bool parameters_applied = media_channel()->SetSendParameters(send_params);
if (!parameters_applied) {
SafeSetError("Failed to set remote audio description send parameters.",
error_desc);
return false;
}
last_send_params_ = send_params;
if (!webrtc::RtpTransceiverDirectionHasSend(content->direction())) {
RTC_DLOG(LS_VERBOSE) << "SetRemoteContent_w: remote side will not send - "
"disable payload type demuxing";
ClearHandledPayloadTypes();
if (!RegisterRtpDemuxerSink()) {
RTC_LOG(LS_ERROR) << "Failed to update audio demuxing.";
return false;
}
}
// TODO(pthatcher): Move remote streams into AudioRecvParameters,
// and only give it to the media channel once we have a local
// description too (without a local description, we won't be able to
// recv them anyway).
if (!UpdateRemoteStreams_w(audio->streams(), type, error_desc)) {
SafeSetError("Failed to set remote audio description streams.", error_desc);
return false;
}
set_remote_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
VideoChannel::VideoChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VideoMediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
UniqueRandomIdGenerator* ssrc_generator)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
std::move(media_channel),
content_name,
srtp_required,
crypto_options,
ssrc_generator) {}
VideoChannel::~VideoChannel() {
TRACE_EVENT0("webrtc", "VideoChannel::~VideoChannel");
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
Deinit();
}
void VideoChannel::UpdateMediaSendRecvState_w() {
// Send outgoing data if we're the active call, we have the remote content,
// and we have had some form of connectivity.
bool send = IsReadyToSendMedia_w();
if (!media_channel()->SetSend(send)) {
RTC_LOG(LS_ERROR) << "Failed to SetSend on video channel";
// TODO(gangji): Report error back to server.
}
RTC_LOG(LS_INFO) << "Changing video state, send=" << send;
}
void VideoChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) {
InvokeOnWorker<void>(RTC_FROM_HERE, Bind(&VideoMediaChannel::FillBitrateInfo,
media_channel(), bwe_info));
}
bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting local video description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find video content in local description.", error_desc);
return false;
}
const VideoContentDescription* video = content->as_video();
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(video->rtp_header_extensions());
UpdateRtpHeaderExtensionMap(rtp_header_extensions);
media_channel()->SetExtmapAllowMixed(video->extmap_allow_mixed());
VideoRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(
video, rtp_header_extensions,