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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
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +000030#include "talk/base/bind.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000031#include "talk/base/buffer.h"
32#include "talk/base/byteorder.h"
33#include "talk/base/common.h"
mallinath@webrtc.org1112c302013-09-23 20:34:45 +000034#include "talk/base/dscp.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000035#include "talk/base/logging.h"
36#include "talk/media/base/rtputils.h"
37#include "talk/p2p/base/transportchannel.h"
38#include "talk/session/media/channelmanager.h"
39#include "talk/session/media/mediamessages.h"
40#include "talk/session/media/rtcpmuxfilter.h"
41#include "talk/session/media/typingmonitor.h"
42
43
44namespace cricket {
45
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +000046using talk_base::Bind;
47
henrike@webrtc.org28e20752013-07-10 00:45:36 +000048enum {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +000049 MSG_EARLYMEDIATIMEOUT = 1,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000050 MSG_SCREENCASTWINDOWEVENT,
51 MSG_RTPPACKET,
52 MSG_RTCPPACKET,
53 MSG_CHANNEL_ERROR,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000054 MSG_READYTOSENDDATA,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000055 MSG_DATARECEIVED,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000056 MSG_FIRSTPACKETRECEIVED,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000057};
58
59// Value specified in RFC 5764.
60static const char kDtlsSrtpExporterLabel[] = "EXTRACTOR-dtls_srtp";
61
62static const int kAgcMinus10db = -10;
63
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000064static void SetSessionError(BaseSession* session, BaseSession::Error error,
65 const std::string& error_desc) {
66 session->SetError(error, error_desc);
67}
68
69static void SafeSetError(const std::string& message, std::string* error_desc) {
70 if (error_desc) {
71 *error_desc = message;
72 }
73}
74
henrike@webrtc.org28e20752013-07-10 00:45:36 +000075// TODO(hellner): use the device manager for creation of screen capturers when
76// the cl enabling it has landed.
77class NullScreenCapturerFactory : public VideoChannel::ScreenCapturerFactory {
78 public:
79 VideoCapturer* CreateScreenCapturer(const ScreencastId& window) {
80 return NULL;
81 }
82};
83
84
85VideoChannel::ScreenCapturerFactory* CreateScreenCapturerFactory() {
86 return new NullScreenCapturerFactory();
87}
88
henrike@webrtc.org28e20752013-07-10 00:45:36 +000089struct PacketMessageData : public talk_base::MessageData {
90 talk_base::Buffer packet;
mallinath@webrtc.org1112c302013-09-23 20:34:45 +000091 talk_base::DiffServCodePoint dscp;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000092};
93
henrike@webrtc.org28e20752013-07-10 00:45:36 +000094struct ScreencastEventMessageData : public talk_base::MessageData {
95 ScreencastEventMessageData(uint32 s, talk_base::WindowEvent we)
96 : ssrc(s),
97 event(we) {
98 }
99 uint32 ssrc;
100 talk_base::WindowEvent event;
101};
102
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000103struct VoiceChannelErrorMessageData : public talk_base::MessageData {
104 VoiceChannelErrorMessageData(uint32 in_ssrc,
105 VoiceMediaChannel::Error in_error)
106 : ssrc(in_ssrc),
107 error(in_error) {
108 }
109 uint32 ssrc;
110 VoiceMediaChannel::Error error;
111};
112
113struct VideoChannelErrorMessageData : public talk_base::MessageData {
114 VideoChannelErrorMessageData(uint32 in_ssrc,
115 VideoMediaChannel::Error in_error)
116 : ssrc(in_ssrc),
117 error(in_error) {
118 }
119 uint32 ssrc;
120 VideoMediaChannel::Error error;
121};
122
123struct DataChannelErrorMessageData : public talk_base::MessageData {
124 DataChannelErrorMessageData(uint32 in_ssrc,
125 DataMediaChannel::Error in_error)
126 : ssrc(in_ssrc),
127 error(in_error) {}
128 uint32 ssrc;
129 DataMediaChannel::Error error;
130};
131
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000132
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000133struct VideoChannel::ScreencastDetailsData {
134 explicit ScreencastDetailsData(uint32 s)
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000135 : ssrc(s), fps(0), screencast_max_pixels(0) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000136 }
137 uint32 ssrc;
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000138 int fps;
139 int screencast_max_pixels;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000140};
141
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000142static const char* PacketType(bool rtcp) {
143 return (!rtcp) ? "RTP" : "RTCP";
144}
145
146static bool ValidPacket(bool rtcp, const talk_base::Buffer* packet) {
147 // Check the packet size. We could check the header too if needed.
148 return (packet &&
149 packet->length() >= (!rtcp ? kMinRtpPacketLen : kMinRtcpPacketLen) &&
150 packet->length() <= kMaxRtpPacketLen);
151}
152
153static bool IsReceiveContentDirection(MediaContentDirection direction) {
154 return direction == MD_SENDRECV || direction == MD_RECVONLY;
155}
156
157static bool IsSendContentDirection(MediaContentDirection direction) {
158 return direction == MD_SENDRECV || direction == MD_SENDONLY;
159}
160
161static const MediaContentDescription* GetContentDescription(
162 const ContentInfo* cinfo) {
163 if (cinfo == NULL)
164 return NULL;
165 return static_cast<const MediaContentDescription*>(cinfo->description);
166}
167
168BaseChannel::BaseChannel(talk_base::Thread* thread,
169 MediaEngineInterface* media_engine,
170 MediaChannel* media_channel, BaseSession* session,
171 const std::string& content_name, bool rtcp)
172 : worker_thread_(thread),
173 media_engine_(media_engine),
174 session_(session),
175 media_channel_(media_channel),
176 content_name_(content_name),
177 rtcp_(rtcp),
178 transport_channel_(NULL),
179 rtcp_transport_channel_(NULL),
180 enabled_(false),
181 writable_(false),
182 rtp_ready_to_send_(false),
183 rtcp_ready_to_send_(false),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000184 was_ever_writable_(false),
185 local_content_direction_(MD_INACTIVE),
186 remote_content_direction_(MD_INACTIVE),
187 has_received_packet_(false),
188 dtls_keyed_(false),
189 secure_required_(false) {
190 ASSERT(worker_thread_ == talk_base::Thread::Current());
191 LOG(LS_INFO) << "Created channel for " << content_name;
192}
193
194BaseChannel::~BaseChannel() {
195 ASSERT(worker_thread_ == talk_base::Thread::Current());
wu@webrtc.org78187522013-10-07 23:32:02 +0000196 Deinit();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000197 StopConnectionMonitor();
198 FlushRtcpMessages(); // Send any outstanding RTCP packets.
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000199 worker_thread_->Clear(this); // eats any outstanding messages or packets
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000200 // We must destroy the media channel before the transport channel, otherwise
201 // the media channel may try to send on the dead transport channel. NULLing
202 // is not an effective strategy since the sends will come on another thread.
203 delete media_channel_;
204 set_rtcp_transport_channel(NULL);
205 if (transport_channel_ != NULL)
206 session_->DestroyChannel(content_name_, transport_channel_->component());
207 LOG(LS_INFO) << "Destroyed channel";
208}
209
210bool BaseChannel::Init(TransportChannel* transport_channel,
211 TransportChannel* rtcp_transport_channel) {
212 if (transport_channel == NULL) {
213 return false;
214 }
215 if (rtcp() && rtcp_transport_channel == NULL) {
216 return false;
217 }
218 transport_channel_ = transport_channel;
219
220 if (!SetDtlsSrtpCiphers(transport_channel_, false)) {
221 return false;
222 }
223
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000224 transport_channel_->SignalWritableState.connect(
225 this, &BaseChannel::OnWritableState);
226 transport_channel_->SignalReadPacket.connect(
227 this, &BaseChannel::OnChannelRead);
228 transport_channel_->SignalReadyToSend.connect(
229 this, &BaseChannel::OnReadyToSend);
230
231 session_->SignalNewLocalDescription.connect(
232 this, &BaseChannel::OnNewLocalDescription);
233 session_->SignalNewRemoteDescription.connect(
234 this, &BaseChannel::OnNewRemoteDescription);
235
236 set_rtcp_transport_channel(rtcp_transport_channel);
wu@webrtc.orgde305012013-10-31 15:40:38 +0000237 // Both RTP and RTCP channels are set, we can call SetInterface on
238 // media channel and it can set network options.
239 media_channel_->SetInterface(this);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000240 return true;
241}
242
wu@webrtc.org78187522013-10-07 23:32:02 +0000243void BaseChannel::Deinit() {
244 media_channel_->SetInterface(NULL);
245}
246
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000247bool BaseChannel::Enable(bool enable) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000248 worker_thread_->Invoke<void>(Bind(
249 enable ? &BaseChannel::EnableMedia_w : &BaseChannel::DisableMedia_w,
250 this));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000251 return true;
252}
253
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000254bool BaseChannel::MuteStream(uint32 ssrc, bool mute) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000255 return InvokeOnWorker(Bind(&BaseChannel::MuteStream_w, this, ssrc, mute));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000256}
257
258bool BaseChannel::IsStreamMuted(uint32 ssrc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000259 return InvokeOnWorker(Bind(&BaseChannel::IsStreamMuted_w, this, ssrc));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000260}
261
262bool BaseChannel::AddRecvStream(const StreamParams& sp) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000263 return InvokeOnWorker(Bind(&BaseChannel::AddRecvStream_w, this, sp));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000264}
265
266bool BaseChannel::RemoveRecvStream(uint32 ssrc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000267 return InvokeOnWorker(Bind(&BaseChannel::RemoveRecvStream_w, this, ssrc));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000268}
269
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000270bool BaseChannel::AddSendStream(const StreamParams& sp) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000271 return InvokeOnWorker(
272 Bind(&MediaChannel::AddSendStream, media_channel(), sp));
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000273}
274
275bool BaseChannel::RemoveSendStream(uint32 ssrc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000276 return InvokeOnWorker(
277 Bind(&MediaChannel::RemoveSendStream, media_channel(), ssrc));
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000278}
279
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000280bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000281 ContentAction action,
282 std::string* error_desc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000283 return InvokeOnWorker(Bind(&BaseChannel::SetLocalContent_w,
284 this, content, action, error_desc));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000285}
286
287bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000288 ContentAction action,
289 std::string* error_desc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000290 return InvokeOnWorker(Bind(&BaseChannel::SetRemoteContent_w,
291 this, content, action, error_desc));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000292}
293
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000294void BaseChannel::StartConnectionMonitor(int cms) {
295 socket_monitor_.reset(new SocketMonitor(transport_channel_,
296 worker_thread(),
297 talk_base::Thread::Current()));
298 socket_monitor_->SignalUpdate.connect(
299 this, &BaseChannel::OnConnectionMonitorUpdate);
300 socket_monitor_->Start(cms);
301}
302
303void BaseChannel::StopConnectionMonitor() {
304 if (socket_monitor_) {
305 socket_monitor_->Stop();
306 socket_monitor_.reset();
307 }
308}
309
310void BaseChannel::set_rtcp_transport_channel(TransportChannel* channel) {
311 if (rtcp_transport_channel_ != channel) {
312 if (rtcp_transport_channel_) {
313 session_->DestroyChannel(
314 content_name_, rtcp_transport_channel_->component());
315 }
316 rtcp_transport_channel_ = channel;
317 if (rtcp_transport_channel_) {
318 // TODO(juberti): Propagate this error code
319 VERIFY(SetDtlsSrtpCiphers(rtcp_transport_channel_, true));
320 rtcp_transport_channel_->SignalWritableState.connect(
321 this, &BaseChannel::OnWritableState);
322 rtcp_transport_channel_->SignalReadPacket.connect(
323 this, &BaseChannel::OnChannelRead);
324 rtcp_transport_channel_->SignalReadyToSend.connect(
325 this, &BaseChannel::OnReadyToSend);
326 }
327 }
328}
329
330bool BaseChannel::IsReadyToReceive() const {
331 // Receive data if we are enabled and have local content,
332 return enabled() && IsReceiveContentDirection(local_content_direction_);
333}
334
335bool BaseChannel::IsReadyToSend() const {
336 // Send outgoing data if we are enabled, have local and remote content,
337 // and we have had some form of connectivity.
338 return enabled() &&
339 IsReceiveContentDirection(remote_content_direction_) &&
340 IsSendContentDirection(local_content_direction_) &&
341 was_ever_writable();
342}
343
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000344bool BaseChannel::SendPacket(talk_base::Buffer* packet,
345 talk_base::DiffServCodePoint dscp) {
346 return SendPacket(false, packet, dscp);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000347}
348
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000349bool BaseChannel::SendRtcp(talk_base::Buffer* packet,
350 talk_base::DiffServCodePoint dscp) {
351 return SendPacket(true, packet, dscp);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000352}
353
354int BaseChannel::SetOption(SocketType type, talk_base::Socket::Option opt,
355 int value) {
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000356 TransportChannel* channel = NULL;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000357 switch (type) {
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000358 case ST_RTP:
359 channel = transport_channel_;
360 break;
361 case ST_RTCP:
362 channel = rtcp_transport_channel_;
363 break;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000364 }
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000365 return channel ? channel->SetOption(opt, value) : -1;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000366}
367
368void BaseChannel::OnWritableState(TransportChannel* channel) {
369 ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
370 if (transport_channel_->writable()
371 && (!rtcp_transport_channel_ || rtcp_transport_channel_->writable())) {
372 ChannelWritable_w();
373 } else {
374 ChannelNotWritable_w();
375 }
376}
377
378void BaseChannel::OnChannelRead(TransportChannel* channel,
wu@webrtc.orga9890802013-12-13 00:21:03 +0000379 const char* data, size_t len,
380 const talk_base::PacketTime& packet_time,
381 int flags) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000382 // OnChannelRead gets called from P2PSocket; now pass data to MediaEngine
383 ASSERT(worker_thread_ == talk_base::Thread::Current());
384
385 // When using RTCP multiplexing we might get RTCP packets on the RTP
386 // transport. We feed RTP traffic into the demuxer to determine if it is RTCP.
387 bool rtcp = PacketIsRtcp(channel, data, len);
388 talk_base::Buffer packet(data, len);
wu@webrtc.orga9890802013-12-13 00:21:03 +0000389 HandlePacket(rtcp, &packet, packet_time);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000390}
391
392void BaseChannel::OnReadyToSend(TransportChannel* channel) {
393 SetReadyToSend(channel, true);
394}
395
396void BaseChannel::SetReadyToSend(TransportChannel* channel, bool ready) {
397 ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
398 if (channel == transport_channel_) {
399 rtp_ready_to_send_ = ready;
400 }
401 if (channel == rtcp_transport_channel_) {
402 rtcp_ready_to_send_ = ready;
403 }
404
405 if (!ready) {
406 // Notify the MediaChannel when either rtp or rtcp channel can't send.
407 media_channel_->OnReadyToSend(false);
408 } else if (rtp_ready_to_send_ &&
409 // In the case of rtcp mux |rtcp_transport_channel_| will be null.
410 (rtcp_ready_to_send_ || !rtcp_transport_channel_)) {
411 // Notify the MediaChannel when both rtp and rtcp channel can send.
412 media_channel_->OnReadyToSend(true);
413 }
414}
415
416bool BaseChannel::PacketIsRtcp(const TransportChannel* channel,
417 const char* data, size_t len) {
418 return (channel == rtcp_transport_channel_ ||
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000419 rtcp_mux_filter_.DemuxRtcp(data, static_cast<int>(len)));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000420}
421
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000422bool BaseChannel::SendPacket(bool rtcp, talk_base::Buffer* packet,
423 talk_base::DiffServCodePoint dscp) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000424 // SendPacket gets called from MediaEngine, typically on an encoder thread.
425 // If the thread is not our worker thread, we will post to our worker
426 // so that the real work happens on our worker. This avoids us having to
427 // synchronize access to all the pieces of the send path, including
428 // SRTP and the inner workings of the transport channels.
429 // The only downside is that we can't return a proper failure code if
430 // needed. Since UDP is unreliable anyway, this should be a non-issue.
431 if (talk_base::Thread::Current() != worker_thread_) {
432 // Avoid a copy by transferring the ownership of the packet data.
433 int message_id = (!rtcp) ? MSG_RTPPACKET : MSG_RTCPPACKET;
434 PacketMessageData* data = new PacketMessageData;
435 packet->TransferTo(&data->packet);
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000436 data->dscp = dscp;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000437 worker_thread_->Post(this, message_id, data);
438 return true;
439 }
440
441 // Now that we are on the correct thread, ensure we have a place to send this
442 // packet before doing anything. (We might get RTCP packets that we don't
443 // intend to send.) If we've negotiated RTCP mux, send RTCP over the RTP
444 // transport.
445 TransportChannel* channel = (!rtcp || rtcp_mux_filter_.IsActive()) ?
446 transport_channel_ : rtcp_transport_channel_;
wu@webrtc.org97077a32013-10-25 21:18:33 +0000447 if (!channel || !channel->writable()) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000448 return false;
449 }
450
451 // Protect ourselves against crazy data.
452 if (!ValidPacket(rtcp, packet)) {
453 LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " "
454 << PacketType(rtcp) << " packet: wrong size="
455 << packet->length();
456 return false;
457 }
458
459 // Signal to the media sink before protecting the packet.
460 {
461 talk_base::CritScope cs(&signal_send_packet_cs_);
462 SignalSendPacketPreCrypto(packet->data(), packet->length(), rtcp);
463 }
464
465 // Protect if needed.
466 if (srtp_filter_.IsActive()) {
467 bool res;
468 char* data = packet->data();
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000469 int len = static_cast<int>(packet->length());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000470 if (!rtcp) {
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000471 res = srtp_filter_.ProtectRtp(data, len,
472 static_cast<int>(packet->capacity()), &len);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000473 if (!res) {
474 int seq_num = -1;
475 uint32 ssrc = 0;
476 GetRtpSeqNum(data, len, &seq_num);
477 GetRtpSsrc(data, len, &ssrc);
478 LOG(LS_ERROR) << "Failed to protect " << content_name_
479 << " RTP packet: size=" << len
480 << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
481 return false;
482 }
483 } else {
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000484 res = srtp_filter_.ProtectRtcp(data, len,
485 static_cast<int>(packet->capacity()),
486 &len);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000487 if (!res) {
488 int type = -1;
489 GetRtcpType(data, len, &type);
490 LOG(LS_ERROR) << "Failed to protect " << content_name_
491 << " RTCP packet: size=" << len << ", type=" << type;
492 return false;
493 }
494 }
495
496 // Update the length of the packet now that we've added the auth tag.
497 packet->SetLength(len);
498 } else if (secure_required_) {
499 // This is a double check for something that supposedly can't happen.
500 LOG(LS_ERROR) << "Can't send outgoing " << PacketType(rtcp)
501 << " packet when SRTP is inactive and crypto is required";
502
503 ASSERT(false);
504 return false;
505 }
506
507 // Signal to the media sink after protecting the packet.
508 {
509 talk_base::CritScope cs(&signal_send_packet_cs_);
510 SignalSendPacketPostCrypto(packet->data(), packet->length(), rtcp);
511 }
512
513 // Bon voyage.
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000514 talk_base::PacketOptions options(dscp);
515 int ret = channel->SendPacket(packet->data(), packet->length(), options,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000516 (secure() && secure_dtls()) ? PF_SRTP_BYPASS : 0);
517 if (ret != static_cast<int>(packet->length())) {
518 if (channel->GetError() == EWOULDBLOCK) {
519 LOG(LS_WARNING) << "Got EWOULDBLOCK from socket.";
520 SetReadyToSend(channel, false);
521 }
522 return false;
523 }
524 return true;
525}
526
527bool BaseChannel::WantsPacket(bool rtcp, talk_base::Buffer* packet) {
528 // Protect ourselves against crazy data.
529 if (!ValidPacket(rtcp, packet)) {
530 LOG(LS_ERROR) << "Dropping incoming " << content_name_ << " "
531 << PacketType(rtcp) << " packet: wrong size="
532 << packet->length();
533 return false;
534 }
535 // If this channel is suppose to handle RTP data, that is determined by
536 // checking against ssrc filter. This is necessary to do it here to avoid
537 // double decryption.
538 if (ssrc_filter_.IsActive() &&
539 !ssrc_filter_.DemuxPacket(packet->data(), packet->length(), rtcp)) {
540 return false;
541 }
542
543 return true;
544}
545
wu@webrtc.orga9890802013-12-13 00:21:03 +0000546void BaseChannel::HandlePacket(bool rtcp, talk_base::Buffer* packet,
547 const talk_base::PacketTime& packet_time) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000548 if (!WantsPacket(rtcp, packet)) {
549 return;
550 }
551
552 if (!has_received_packet_) {
553 has_received_packet_ = true;
554 signaling_thread()->Post(this, MSG_FIRSTPACKETRECEIVED);
555 }
556
557 // Signal to the media sink before unprotecting the packet.
558 {
559 talk_base::CritScope cs(&signal_recv_packet_cs_);
560 SignalRecvPacketPostCrypto(packet->data(), packet->length(), rtcp);
561 }
562
563 // Unprotect the packet, if needed.
564 if (srtp_filter_.IsActive()) {
565 char* data = packet->data();
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000566 int len = static_cast<int>(packet->length());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000567 bool res;
568 if (!rtcp) {
569 res = srtp_filter_.UnprotectRtp(data, len, &len);
570 if (!res) {
571 int seq_num = -1;
572 uint32 ssrc = 0;
573 GetRtpSeqNum(data, len, &seq_num);
574 GetRtpSsrc(data, len, &ssrc);
575 LOG(LS_ERROR) << "Failed to unprotect " << content_name_
576 << " RTP packet: size=" << len
577 << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
578 return;
579 }
580 } else {
581 res = srtp_filter_.UnprotectRtcp(data, len, &len);
582 if (!res) {
583 int type = -1;
584 GetRtcpType(data, len, &type);
585 LOG(LS_ERROR) << "Failed to unprotect " << content_name_
586 << " RTCP packet: size=" << len << ", type=" << type;
587 return;
588 }
589 }
590
591 packet->SetLength(len);
592 } else if (secure_required_) {
593 // Our session description indicates that SRTP is required, but we got a
594 // packet before our SRTP filter is active. This means either that
595 // a) we got SRTP packets before we received the SDES keys, in which case
596 // we can't decrypt it anyway, or
597 // b) we got SRTP packets before DTLS completed on both the RTP and RTCP
598 // channels, so we haven't yet extracted keys, even if DTLS did complete
599 // on the channel that the packets are being sent on. It's really good
600 // practice to wait for both RTP and RTCP to be good to go before sending
601 // media, to prevent weird failure modes, so it's fine for us to just eat
602 // packets here. This is all sidestepped if RTCP mux is used anyway.
603 LOG(LS_WARNING) << "Can't process incoming " << PacketType(rtcp)
604 << " packet when SRTP is inactive and crypto is required";
605 return;
606 }
607
608 // Signal to the media sink after unprotecting the packet.
609 {
610 talk_base::CritScope cs(&signal_recv_packet_cs_);
611 SignalRecvPacketPreCrypto(packet->data(), packet->length(), rtcp);
612 }
613
614 // Push it down to the media channel.
615 if (!rtcp) {
wu@webrtc.orga9890802013-12-13 00:21:03 +0000616 media_channel_->OnPacketReceived(packet, packet_time);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000617 } else {
wu@webrtc.orga9890802013-12-13 00:21:03 +0000618 media_channel_->OnRtcpReceived(packet, packet_time);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000619 }
620}
621
622void BaseChannel::OnNewLocalDescription(
623 BaseSession* session, ContentAction action) {
624 const ContentInfo* content_info =
625 GetFirstContent(session->local_description());
626 const MediaContentDescription* content_desc =
627 GetContentDescription(content_info);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000628 std::string error_desc;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000629 if (content_desc && content_info && !content_info->rejected &&
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000630 !SetLocalContent(content_desc, action, &error_desc)) {
631 SetSessionError(session_, BaseSession::ERROR_CONTENT, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000632 LOG(LS_ERROR) << "Failure in SetLocalContent with action " << action;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000633 }
634}
635
636void BaseChannel::OnNewRemoteDescription(
637 BaseSession* session, ContentAction action) {
638 const ContentInfo* content_info =
639 GetFirstContent(session->remote_description());
640 const MediaContentDescription* content_desc =
641 GetContentDescription(content_info);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000642 std::string error_desc;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000643 if (content_desc && content_info && !content_info->rejected &&
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000644 !SetRemoteContent(content_desc, action, &error_desc)) {
645 SetSessionError(session_, BaseSession::ERROR_CONTENT, error_desc);
646 LOG(LS_ERROR) << "Failure in SetRemoteContent with action " << action;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000647 }
648}
649
650void BaseChannel::EnableMedia_w() {
651 ASSERT(worker_thread_ == talk_base::Thread::Current());
652 if (enabled_)
653 return;
654
655 LOG(LS_INFO) << "Channel enabled";
656 enabled_ = true;
657 ChangeState();
658}
659
660void BaseChannel::DisableMedia_w() {
661 ASSERT(worker_thread_ == talk_base::Thread::Current());
662 if (!enabled_)
663 return;
664
665 LOG(LS_INFO) << "Channel disabled";
666 enabled_ = false;
667 ChangeState();
668}
669
670bool BaseChannel::MuteStream_w(uint32 ssrc, bool mute) {
671 ASSERT(worker_thread_ == talk_base::Thread::Current());
672 bool ret = media_channel()->MuteStream(ssrc, mute);
673 if (ret) {
674 if (mute)
675 muted_streams_.insert(ssrc);
676 else
677 muted_streams_.erase(ssrc);
678 }
679 return ret;
680}
681
682bool BaseChannel::IsStreamMuted_w(uint32 ssrc) {
683 ASSERT(worker_thread_ == talk_base::Thread::Current());
684 return muted_streams_.find(ssrc) != muted_streams_.end();
685}
686
687void BaseChannel::ChannelWritable_w() {
688 ASSERT(worker_thread_ == talk_base::Thread::Current());
689 if (writable_)
690 return;
691
692 LOG(LS_INFO) << "Channel socket writable ("
693 << transport_channel_->content_name() << ", "
694 << transport_channel_->component() << ")"
695 << (was_ever_writable_ ? "" : " for the first time");
696
697 std::vector<ConnectionInfo> infos;
698 transport_channel_->GetStats(&infos);
699 for (std::vector<ConnectionInfo>::const_iterator it = infos.begin();
700 it != infos.end(); ++it) {
701 if (it->best_connection) {
702 LOG(LS_INFO) << "Using " << it->local_candidate.ToSensitiveString()
703 << "->" << it->remote_candidate.ToSensitiveString();
704 break;
705 }
706 }
707
708 // If we're doing DTLS-SRTP, now is the time.
709 if (!was_ever_writable_ && ShouldSetupDtlsSrtp()) {
710 if (!SetupDtlsSrtp(false)) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000711 const std::string error_desc =
712 "Couldn't set up DTLS-SRTP on RTP channel.";
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000713 // Sent synchronously.
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000714 signaling_thread()->Invoke<void>(Bind(
715 &SetSessionError,
716 session_,
717 BaseSession::ERROR_TRANSPORT,
718 error_desc));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000719 return;
720 }
721
722 if (rtcp_transport_channel_) {
723 if (!SetupDtlsSrtp(true)) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000724 const std::string error_desc =
725 "Couldn't set up DTLS-SRTP on RTCP channel";
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000726 // Sent synchronously.
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000727 signaling_thread()->Invoke<void>(Bind(
728 &SetSessionError,
729 session_,
730 BaseSession::ERROR_TRANSPORT,
731 error_desc));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000732 return;
733 }
734 }
735 }
736
737 was_ever_writable_ = true;
738 writable_ = true;
739 ChangeState();
740}
741
742bool BaseChannel::SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp) {
743 std::vector<std::string> ciphers;
744 // We always use the default SRTP ciphers for RTCP, but we may use different
745 // ciphers for RTP depending on the media type.
746 if (!rtcp) {
747 GetSrtpCiphers(&ciphers);
748 } else {
749 GetSupportedDefaultCryptoSuites(&ciphers);
750 }
751 return tc->SetSrtpCiphers(ciphers);
752}
753
754bool BaseChannel::ShouldSetupDtlsSrtp() const {
755 return true;
756}
757
758// This function returns true if either DTLS-SRTP is not in use
759// *or* DTLS-SRTP is successfully set up.
760bool BaseChannel::SetupDtlsSrtp(bool rtcp_channel) {
761 bool ret = false;
762
763 TransportChannel *channel = rtcp_channel ?
764 rtcp_transport_channel_ : transport_channel_;
765
766 // No DTLS
767 if (!channel->IsDtlsActive())
768 return true;
769
770 std::string selected_cipher;
771
772 if (!channel->GetSrtpCipher(&selected_cipher)) {
773 LOG(LS_ERROR) << "No DTLS-SRTP selected cipher";
774 return false;
775 }
776
777 LOG(LS_INFO) << "Installing keys from DTLS-SRTP on "
778 << content_name() << " "
779 << PacketType(rtcp_channel);
780
781 // OK, we're now doing DTLS (RFC 5764)
782 std::vector<unsigned char> dtls_buffer(SRTP_MASTER_KEY_KEY_LEN * 2 +
783 SRTP_MASTER_KEY_SALT_LEN * 2);
784
785 // RFC 5705 exporter using the RFC 5764 parameters
786 if (!channel->ExportKeyingMaterial(
787 kDtlsSrtpExporterLabel,
788 NULL, 0, false,
789 &dtls_buffer[0], dtls_buffer.size())) {
790 LOG(LS_WARNING) << "DTLS-SRTP key export failed";
791 ASSERT(false); // This should never happen
792 return false;
793 }
794
795 // Sync up the keys with the DTLS-SRTP interface
796 std::vector<unsigned char> client_write_key(SRTP_MASTER_KEY_KEY_LEN +
797 SRTP_MASTER_KEY_SALT_LEN);
798 std::vector<unsigned char> server_write_key(SRTP_MASTER_KEY_KEY_LEN +
799 SRTP_MASTER_KEY_SALT_LEN);
800 size_t offset = 0;
801 memcpy(&client_write_key[0], &dtls_buffer[offset],
802 SRTP_MASTER_KEY_KEY_LEN);
803 offset += SRTP_MASTER_KEY_KEY_LEN;
804 memcpy(&server_write_key[0], &dtls_buffer[offset],
805 SRTP_MASTER_KEY_KEY_LEN);
806 offset += SRTP_MASTER_KEY_KEY_LEN;
807 memcpy(&client_write_key[SRTP_MASTER_KEY_KEY_LEN],
808 &dtls_buffer[offset], SRTP_MASTER_KEY_SALT_LEN);
809 offset += SRTP_MASTER_KEY_SALT_LEN;
810 memcpy(&server_write_key[SRTP_MASTER_KEY_KEY_LEN],
811 &dtls_buffer[offset], SRTP_MASTER_KEY_SALT_LEN);
812
813 std::vector<unsigned char> *send_key, *recv_key;
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000814 talk_base::SSLRole role;
815 if (!channel->GetSslRole(&role)) {
816 LOG(LS_WARNING) << "GetSslRole failed";
817 return false;
818 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000819
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000820 if (role == talk_base::SSL_SERVER) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000821 send_key = &server_write_key;
822 recv_key = &client_write_key;
823 } else {
824 send_key = &client_write_key;
825 recv_key = &server_write_key;
826 }
827
828 if (rtcp_channel) {
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000829 ret = srtp_filter_.SetRtcpParams(
830 selected_cipher,
831 &(*send_key)[0],
832 static_cast<int>(send_key->size()),
833 selected_cipher,
834 &(*recv_key)[0],
835 static_cast<int>(recv_key->size()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000836 } else {
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000837 ret = srtp_filter_.SetRtpParams(
838 selected_cipher,
839 &(*send_key)[0],
840 static_cast<int>(send_key->size()),
841 selected_cipher,
842 &(*recv_key)[0],
843 static_cast<int>(recv_key->size()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000844 }
845
846 if (!ret)
847 LOG(LS_WARNING) << "DTLS-SRTP key installation failed";
848 else
849 dtls_keyed_ = true;
850
851 return ret;
852}
853
854void BaseChannel::ChannelNotWritable_w() {
855 ASSERT(worker_thread_ == talk_base::Thread::Current());
856 if (!writable_)
857 return;
858
859 LOG(LS_INFO) << "Channel socket not writable ("
860 << transport_channel_->content_name() << ", "
861 << transport_channel_->component() << ")";
862 writable_ = false;
863 ChangeState();
864}
865
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000866// |dtls| will be set to true if DTLS is active for transport channel and
867// crypto is empty.
868bool BaseChannel::CheckSrtpConfig(const std::vector<CryptoParams>& cryptos,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000869 bool* dtls,
870 std::string* error_desc) {
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000871 *dtls = transport_channel_->IsDtlsActive();
872 if (*dtls && !cryptos.empty()) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000873 SafeSetError("Cryptos must be empty when DTLS is active.",
874 error_desc);
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000875 return false;
876 }
877 return true;
878}
879
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000880bool BaseChannel::SetSrtp_w(const std::vector<CryptoParams>& cryptos,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000881 ContentAction action,
882 ContentSource src,
883 std::string* error_desc) {
884 if (action == CA_UPDATE) {
885 // no crypto params.
886 return true;
887 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000888 bool ret = false;
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000889 bool dtls = false;
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000890 ret = CheckSrtpConfig(cryptos, &dtls, error_desc);
891 if (!ret) {
892 return false;
893 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000894 switch (action) {
895 case CA_OFFER:
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000896 // If DTLS is already active on the channel, we could be renegotiating
897 // here. We don't update the srtp filter.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000898 if (!dtls) {
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000899 ret = srtp_filter_.SetOffer(cryptos, src);
900 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000901 break;
902 case CA_PRANSWER:
903 // If we're doing DTLS-SRTP, we don't want to update the filter
904 // with an answer, because we already have SRTP parameters.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000905 if (!dtls) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000906 ret = srtp_filter_.SetProvisionalAnswer(cryptos, src);
907 }
908 break;
909 case CA_ANSWER:
910 // If we're doing DTLS-SRTP, we don't want to update the filter
911 // with an answer, because we already have SRTP parameters.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000912 if (!dtls) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000913 ret = srtp_filter_.SetAnswer(cryptos, src);
914 }
915 break;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000916 default:
917 break;
918 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000919 if (!ret) {
920 SafeSetError("Failed to setup SRTP filter.", error_desc);
921 return false;
922 }
923 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000924}
925
926bool BaseChannel::SetRtcpMux_w(bool enable, ContentAction action,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000927 ContentSource src,
928 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000929 bool ret = false;
930 switch (action) {
931 case CA_OFFER:
932 ret = rtcp_mux_filter_.SetOffer(enable, src);
933 break;
934 case CA_PRANSWER:
935 ret = rtcp_mux_filter_.SetProvisionalAnswer(enable, src);
936 break;
937 case CA_ANSWER:
938 ret = rtcp_mux_filter_.SetAnswer(enable, src);
939 if (ret && rtcp_mux_filter_.IsActive()) {
940 // We activated RTCP mux, close down the RTCP transport.
941 set_rtcp_transport_channel(NULL);
942 }
943 break;
944 case CA_UPDATE:
945 // No RTCP mux info.
946 ret = true;
947 default:
948 break;
949 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000950 if (!ret) {
951 SafeSetError("Failed to setup RTCP mux filter.", error_desc);
952 return false;
953 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000954 // |rtcp_mux_filter_| can be active if |action| is CA_PRANSWER or
955 // CA_ANSWER, but we only want to tear down the RTCP transport channel if we
956 // received a final answer.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000957 if (rtcp_mux_filter_.IsActive()) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000958 // If the RTP transport is already writable, then so are we.
959 if (transport_channel_->writable()) {
960 ChannelWritable_w();
961 }
962 }
963
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000964 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000965}
966
967bool BaseChannel::AddRecvStream_w(const StreamParams& sp) {
968 ASSERT(worker_thread() == talk_base::Thread::Current());
969 if (!media_channel()->AddRecvStream(sp))
970 return false;
971
972 return ssrc_filter_.AddStream(sp);
973}
974
975bool BaseChannel::RemoveRecvStream_w(uint32 ssrc) {
976 ASSERT(worker_thread() == talk_base::Thread::Current());
977 ssrc_filter_.RemoveStream(ssrc);
978 return media_channel()->RemoveRecvStream(ssrc);
979}
980
981bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000982 ContentAction action,
983 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000984 if (!VERIFY(action == CA_OFFER || action == CA_ANSWER ||
985 action == CA_PRANSWER || action == CA_UPDATE))
986 return false;
987
988 // If this is an update, streams only contain streams that have changed.
989 if (action == CA_UPDATE) {
990 for (StreamParamsVec::const_iterator it = streams.begin();
991 it != streams.end(); ++it) {
992 StreamParams existing_stream;
993 bool stream_exist = GetStreamByIds(local_streams_, it->groupid,
994 it->id, &existing_stream);
995 if (!stream_exist && it->has_ssrcs()) {
996 if (media_channel()->AddSendStream(*it)) {
997 local_streams_.push_back(*it);
998 LOG(LS_INFO) << "Add send stream ssrc: " << it->first_ssrc();
999 } else {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001000 std::ostringstream desc;
1001 desc << "Failed to add send stream ssrc: " << it->first_ssrc();
1002 SafeSetError(desc.str(), error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001003 return false;
1004 }
1005 } else if (stream_exist && !it->has_ssrcs()) {
1006 if (!media_channel()->RemoveSendStream(existing_stream.first_ssrc())) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001007 std::ostringstream desc;
1008 desc << "Failed to remove send stream with ssrc "
1009 << it->first_ssrc() << ".";
1010 SafeSetError(desc.str(), error_desc);
1011 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001012 }
1013 RemoveStreamBySsrc(&local_streams_, existing_stream.first_ssrc());
1014 } else {
1015 LOG(LS_WARNING) << "Ignore unsupported stream update";
1016 }
1017 }
1018 return true;
1019 }
1020 // Else streams are all the streams we want to send.
1021
1022 // Check for streams that have been removed.
1023 bool ret = true;
1024 for (StreamParamsVec::const_iterator it = local_streams_.begin();
1025 it != local_streams_.end(); ++it) {
1026 if (!GetStreamBySsrc(streams, it->first_ssrc(), NULL)) {
1027 if (!media_channel()->RemoveSendStream(it->first_ssrc())) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001028 std::ostringstream desc;
1029 desc << "Failed to remove send stream with ssrc "
1030 << it->first_ssrc() << ".";
1031 SafeSetError(desc.str(), error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001032 ret = false;
1033 }
1034 }
1035 }
1036 // Check for new streams.
1037 for (StreamParamsVec::const_iterator it = streams.begin();
1038 it != streams.end(); ++it) {
1039 if (!GetStreamBySsrc(local_streams_, it->first_ssrc(), NULL)) {
1040 if (media_channel()->AddSendStream(*it)) {
1041 LOG(LS_INFO) << "Add send ssrc: " << it->ssrcs[0];
1042 } else {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001043 std::ostringstream desc;
1044 desc << "Failed to add send stream ssrc: " << it->first_ssrc();
1045 SafeSetError(desc.str(), error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001046 ret = false;
1047 }
1048 }
1049 }
1050 local_streams_ = streams;
1051 return ret;
1052}
1053
1054bool BaseChannel::UpdateRemoteStreams_w(
1055 const std::vector<StreamParams>& streams,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001056 ContentAction action,
1057 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001058 if (!VERIFY(action == CA_OFFER || action == CA_ANSWER ||
1059 action == CA_PRANSWER || action == CA_UPDATE))
1060 return false;
1061
1062 // If this is an update, streams only contain streams that have changed.
1063 if (action == CA_UPDATE) {
1064 for (StreamParamsVec::const_iterator it = streams.begin();
1065 it != streams.end(); ++it) {
1066 StreamParams existing_stream;
1067 bool stream_exists = GetStreamByIds(remote_streams_, it->groupid,
1068 it->id, &existing_stream);
1069 if (!stream_exists && it->has_ssrcs()) {
1070 if (AddRecvStream_w(*it)) {
1071 remote_streams_.push_back(*it);
1072 LOG(LS_INFO) << "Add remote stream ssrc: " << it->first_ssrc();
1073 } else {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001074 std::ostringstream desc;
1075 desc << "Failed to add remote stream ssrc: " << it->first_ssrc();
1076 SafeSetError(desc.str(), error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001077 return false;
1078 }
1079 } else if (stream_exists && !it->has_ssrcs()) {
1080 if (!RemoveRecvStream_w(existing_stream.first_ssrc())) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001081 std::ostringstream desc;
1082 desc << "Failed to remove remote stream with ssrc "
1083 << it->first_ssrc() << ".";
1084 SafeSetError(desc.str(), error_desc);
1085 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001086 }
1087 RemoveStreamBySsrc(&remote_streams_, existing_stream.first_ssrc());
1088 } else {
1089 LOG(LS_WARNING) << "Ignore unsupported stream update."
1090 << " Stream exists? " << stream_exists
1091 << " existing stream = " << existing_stream.ToString()
1092 << " new stream = " << it->ToString();
1093 }
1094 }
1095 return true;
1096 }
1097 // Else streams are all the streams we want to receive.
1098
1099 // Check for streams that have been removed.
1100 bool ret = true;
1101 for (StreamParamsVec::const_iterator it = remote_streams_.begin();
1102 it != remote_streams_.end(); ++it) {
1103 if (!GetStreamBySsrc(streams, it->first_ssrc(), NULL)) {
1104 if (!RemoveRecvStream_w(it->first_ssrc())) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001105 std::ostringstream desc;
1106 desc << "Failed to remove remote stream with ssrc "
1107 << it->first_ssrc() << ".";
1108 SafeSetError(desc.str(), error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001109 ret = false;
1110 }
1111 }
1112 }
1113 // Check for new streams.
1114 for (StreamParamsVec::const_iterator it = streams.begin();
1115 it != streams.end(); ++it) {
1116 if (!GetStreamBySsrc(remote_streams_, it->first_ssrc(), NULL)) {
1117 if (AddRecvStream_w(*it)) {
1118 LOG(LS_INFO) << "Add remote ssrc: " << it->ssrcs[0];
1119 } else {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001120 std::ostringstream desc;
1121 desc << "Failed to add remote stream ssrc: " << it->first_ssrc();
1122 SafeSetError(desc.str(), error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001123 ret = false;
1124 }
1125 }
1126 }
1127 remote_streams_ = streams;
1128 return ret;
1129}
1130
1131bool BaseChannel::SetBaseLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001132 ContentAction action,
1133 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001134 // Cache secure_required_ for belt and suspenders check on SendPacket
1135 secure_required_ = content->crypto_required();
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001136 bool ret = UpdateLocalStreams_w(content->streams(), action, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001137 // Set local SRTP parameters (what we will encrypt with).
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001138 ret &= SetSrtp_w(content->cryptos(), action, CS_LOCAL, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001139 // Set local RTCP mux parameters.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001140 ret &= SetRtcpMux_w(content->rtcp_mux(), action, CS_LOCAL, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001141 // Set local RTP header extensions.
1142 if (content->rtp_header_extensions_set()) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001143 if (!media_channel()->SetRecvRtpHeaderExtensions(
1144 content->rtp_header_extensions())) {
1145 std::ostringstream desc;
1146 desc << "Failed to set receive rtp header extensions for "
1147 << MediaTypeToString(content->type()) << " content.";
1148 SafeSetError(desc.str(), error_desc);
1149 ret = false;
1150 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001151 }
1152 set_local_content_direction(content->direction());
1153 return ret;
1154}
1155
1156bool BaseChannel::SetBaseRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001157 ContentAction action,
1158 std::string* error_desc) {
1159 bool ret = UpdateRemoteStreams_w(content->streams(), action, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001160 // Set remote SRTP parameters (what the other side will encrypt with).
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001161 ret &= SetSrtp_w(content->cryptos(), action, CS_REMOTE, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001162 // Set remote RTCP mux parameters.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001163 ret &= SetRtcpMux_w(content->rtcp_mux(), action, CS_REMOTE, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001164 // Set remote RTP header extensions.
1165 if (content->rtp_header_extensions_set()) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001166 if (!media_channel()->SetSendRtpHeaderExtensions(
1167 content->rtp_header_extensions())) {
1168 std::ostringstream desc;
1169 desc << "Failed to set send rtp header extensions for "
1170 << MediaTypeToString(content->type()) << " content.";
1171 SafeSetError(desc.str(), error_desc);
1172 ret = false;
1173 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001174 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001175
1176 if (!media_channel()->SetMaxSendBandwidth(content->bandwidth())) {
1177 std::ostringstream desc;
1178 desc << "Failed to set max send bandwidth for "
1179 << MediaTypeToString(content->type()) << " content.";
1180 SafeSetError(desc.str(), error_desc);
1181 ret = false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001182 }
1183 set_remote_content_direction(content->direction());
1184 return ret;
1185}
1186
1187void BaseChannel::OnMessage(talk_base::Message *pmsg) {
1188 switch (pmsg->message_id) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001189 case MSG_RTPPACKET:
1190 case MSG_RTCPPACKET: {
1191 PacketMessageData* data = static_cast<PacketMessageData*>(pmsg->pdata);
mallinath@webrtc.org1112c302013-09-23 20:34:45 +00001192 SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet, data->dscp);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001193 delete data; // because it is Posted
1194 break;
1195 }
1196 case MSG_FIRSTPACKETRECEIVED: {
1197 SignalFirstPacketReceived(this);
1198 break;
1199 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001200 }
1201}
1202
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001203void BaseChannel::FlushRtcpMessages() {
1204 // Flush all remaining RTCP messages. This should only be called in
1205 // destructor.
1206 ASSERT(talk_base::Thread::Current() == worker_thread_);
1207 talk_base::MessageList rtcp_messages;
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001208 worker_thread_->Clear(this, MSG_RTCPPACKET, &rtcp_messages);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001209 for (talk_base::MessageList::iterator it = rtcp_messages.begin();
1210 it != rtcp_messages.end(); ++it) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001211 worker_thread_->Send(this, MSG_RTCPPACKET, it->pdata);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001212 }
1213}
1214
1215VoiceChannel::VoiceChannel(talk_base::Thread* thread,
1216 MediaEngineInterface* media_engine,
1217 VoiceMediaChannel* media_channel,
1218 BaseSession* session,
1219 const std::string& content_name,
1220 bool rtcp)
1221 : BaseChannel(thread, media_engine, media_channel, session, content_name,
1222 rtcp),
1223 received_media_(false) {
1224}
1225
1226VoiceChannel::~VoiceChannel() {
1227 StopAudioMonitor();
1228 StopMediaMonitor();
1229 // this can't be done in the base class, since it calls a virtual
1230 DisableMedia_w();
wu@webrtc.org78187522013-10-07 23:32:02 +00001231 Deinit();
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001232}
1233
1234bool VoiceChannel::Init() {
1235 TransportChannel* rtcp_channel = rtcp() ? session()->CreateChannel(
1236 content_name(), "rtcp", ICE_CANDIDATE_COMPONENT_RTCP) : NULL;
1237 if (!BaseChannel::Init(session()->CreateChannel(
1238 content_name(), "rtp", ICE_CANDIDATE_COMPONENT_RTP),
1239 rtcp_channel)) {
1240 return false;
1241 }
1242 media_channel()->SignalMediaError.connect(
1243 this, &VoiceChannel::OnVoiceChannelError);
1244 srtp_filter()->SignalSrtpError.connect(
1245 this, &VoiceChannel::OnSrtpError);
1246 return true;
1247}
1248
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00001249bool VoiceChannel::SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001250 return InvokeOnWorker(Bind(&VoiceMediaChannel::SetRemoteRenderer,
1251 media_channel(), ssrc, renderer));
henrike@webrtc.org1e09a712013-07-26 19:17:59 +00001252}
1253
1254bool VoiceChannel::SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001255 return InvokeOnWorker(Bind(&VoiceMediaChannel::SetLocalRenderer,
1256 media_channel(), ssrc, renderer));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001257}
1258
1259bool VoiceChannel::SetRingbackTone(const void* buf, int len) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001260 return InvokeOnWorker(Bind(&VoiceChannel::SetRingbackTone_w, this, buf, len));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001261}
1262
1263// TODO(juberti): Handle early media the right way. We should get an explicit
1264// ringing message telling us to start playing local ringback, which we cancel
1265// if any early media actually arrives. For now, we do the opposite, which is
1266// to wait 1 second for early media, and start playing local ringback if none
1267// arrives.
1268void VoiceChannel::SetEarlyMedia(bool enable) {
1269 if (enable) {
1270 // Start the early media timeout
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001271 worker_thread()->PostDelayed(kEarlyMediaTimeout, this,
1272 MSG_EARLYMEDIATIMEOUT);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001273 } else {
1274 // Stop the timeout if currently going.
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001275 worker_thread()->Clear(this, MSG_EARLYMEDIATIMEOUT);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001276 }
1277}
1278
1279bool VoiceChannel::PlayRingbackTone(uint32 ssrc, bool play, bool loop) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001280 return InvokeOnWorker(Bind(&VoiceChannel::PlayRingbackTone_w,
1281 this, ssrc, play, loop));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001282}
1283
1284bool VoiceChannel::PressDTMF(int digit, bool playout) {
1285 int flags = DF_SEND;
1286 if (playout) {
1287 flags |= DF_PLAY;
1288 }
1289 int duration_ms = 160;
1290 return InsertDtmf(0, digit, duration_ms, flags);
1291}
1292
1293bool VoiceChannel::CanInsertDtmf() {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001294 return InvokeOnWorker(Bind(&VoiceMediaChannel::CanInsertDtmf,
1295 media_channel()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001296}
1297
1298bool VoiceChannel::InsertDtmf(uint32 ssrc, int event_code, int duration,
1299 int flags) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001300 return InvokeOnWorker(Bind(&VoiceChannel::InsertDtmf_w, this,
1301 ssrc, event_code, duration, flags));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001302}
1303
1304bool VoiceChannel::SetOutputScaling(uint32 ssrc, double left, double right) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001305 return InvokeOnWorker(Bind(&VoiceMediaChannel::SetOutputScaling,
1306 media_channel(), ssrc, left, right));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001307}
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001308
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001309bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001310 return InvokeOnWorker(Bind(&VoiceMediaChannel::GetStats,
1311 media_channel(), stats));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001312}
1313
1314void VoiceChannel::StartMediaMonitor(int cms) {
1315 media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(),
1316 talk_base::Thread::Current()));
1317 media_monitor_->SignalUpdate.connect(
1318 this, &VoiceChannel::OnMediaMonitorUpdate);
1319 media_monitor_->Start(cms);
1320}
1321
1322void VoiceChannel::StopMediaMonitor() {
1323 if (media_monitor_) {
1324 media_monitor_->Stop();
1325 media_monitor_->SignalUpdate.disconnect(this);
1326 media_monitor_.reset();
1327 }
1328}
1329
1330void VoiceChannel::StartAudioMonitor(int cms) {
1331 audio_monitor_.reset(new AudioMonitor(this, talk_base::Thread::Current()));
1332 audio_monitor_
1333 ->SignalUpdate.connect(this, &VoiceChannel::OnAudioMonitorUpdate);
1334 audio_monitor_->Start(cms);
1335}
1336
1337void VoiceChannel::StopAudioMonitor() {
1338 if (audio_monitor_) {
1339 audio_monitor_->Stop();
1340 audio_monitor_.reset();
1341 }
1342}
1343
1344bool VoiceChannel::IsAudioMonitorRunning() const {
1345 return (audio_monitor_.get() != NULL);
1346}
1347
1348void VoiceChannel::StartTypingMonitor(const TypingMonitorOptions& settings) {
1349 typing_monitor_.reset(new TypingMonitor(this, worker_thread(), settings));
1350 SignalAutoMuted.repeat(typing_monitor_->SignalMuted);
1351}
1352
1353void VoiceChannel::StopTypingMonitor() {
1354 typing_monitor_.reset();
1355}
1356
1357bool VoiceChannel::IsTypingMonitorRunning() const {
1358 return typing_monitor_;
1359}
1360
1361bool VoiceChannel::MuteStream_w(uint32 ssrc, bool mute) {
1362 bool ret = BaseChannel::MuteStream_w(ssrc, mute);
1363 if (typing_monitor_ && mute)
1364 typing_monitor_->OnChannelMuted();
1365 return ret;
1366}
1367
1368int VoiceChannel::GetInputLevel_w() {
1369 return media_engine()->GetInputLevel();
1370}
1371
1372int VoiceChannel::GetOutputLevel_w() {
1373 return media_channel()->GetOutputLevel();
1374}
1375
1376void VoiceChannel::GetActiveStreams_w(AudioInfo::StreamList* actives) {
1377 media_channel()->GetActiveStreams(actives);
1378}
1379
1380void VoiceChannel::OnChannelRead(TransportChannel* channel,
wu@webrtc.orga9890802013-12-13 00:21:03 +00001381 const char* data, size_t len,
1382 const talk_base::PacketTime& packet_time,
1383 int flags) {
1384 BaseChannel::OnChannelRead(channel, data, len, packet_time, flags);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001385
1386 // Set a flag when we've received an RTP packet. If we're waiting for early
1387 // media, this will disable the timeout.
1388 if (!received_media_ && !PacketIsRtcp(channel, data, len)) {
1389 received_media_ = true;
1390 }
1391}
1392
1393void VoiceChannel::ChangeState() {
1394 // Render incoming data if we're the active call, and we have the local
1395 // content. We receive data on the default channel and multiplexed streams.
1396 bool recv = IsReadyToReceive();
1397 if (!media_channel()->SetPlayout(recv)) {
1398 SendLastMediaError();
1399 }
1400
1401 // Send outgoing data if we're the active call, we have the remote content,
1402 // and we have had some form of connectivity.
1403 bool send = IsReadyToSend();
1404 SendFlags send_flag = send ? SEND_MICROPHONE : SEND_NOTHING;
1405 if (!media_channel()->SetSend(send_flag)) {
1406 LOG(LS_ERROR) << "Failed to SetSend " << send_flag << " on voice channel";
1407 SendLastMediaError();
1408 }
1409
1410 LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send;
1411}
1412
1413const ContentInfo* VoiceChannel::GetFirstContent(
1414 const SessionDescription* sdesc) {
1415 return GetFirstAudioContent(sdesc);
1416}
1417
1418bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001419 ContentAction action,
1420 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001421 ASSERT(worker_thread() == talk_base::Thread::Current());
1422 LOG(LS_INFO) << "Setting local voice description";
1423
1424 const AudioContentDescription* audio =
1425 static_cast<const AudioContentDescription*>(content);
1426 ASSERT(audio != NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001427 if (!audio) {
1428 SafeSetError("Can't find audio content in local description.", error_desc);
1429 return false;
1430 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001431
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001432 bool ret = SetBaseLocalContent_w(content, action, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001433 // Set local audio codecs (what we want to receive).
1434 // TODO(whyuan): Change action != CA_UPDATE to !audio->partial() when partial
1435 // is set properly.
1436 if (action != CA_UPDATE || audio->has_codecs()) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001437 if (!media_channel()->SetRecvCodecs(audio->codecs())) {
1438 SafeSetError("Failed to set audio receive codecs.", error_desc);
1439 ret = false;
1440 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001441 }
1442
1443 // If everything worked, see if we can start receiving.
1444 if (ret) {
1445 ChangeState();
1446 } else {
1447 LOG(LS_WARNING) << "Failed to set local voice description";
1448 }
1449 return ret;
1450}
1451
1452bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001453 ContentAction action,
1454 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001455 ASSERT(worker_thread() == talk_base::Thread::Current());
1456 LOG(LS_INFO) << "Setting remote voice description";
1457
1458 const AudioContentDescription* audio =
1459 static_cast<const AudioContentDescription*>(content);
1460 ASSERT(audio != NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001461 if (!audio) {
1462 SafeSetError("Can't find audio content in remote description.", error_desc);
1463 return false;
1464 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001465
1466 bool ret = true;
1467 // Set remote video codecs (what the other side wants to receive).
1468 if (action != CA_UPDATE || audio->has_codecs()) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001469 if (!media_channel()->SetSendCodecs(audio->codecs())) {
1470 SafeSetError("Failed to set audio send codecs.", error_desc);
1471 ret = false;
1472 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001473 }
1474
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001475 ret &= SetBaseRemoteContent_w(content, action, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001476
1477 if (action != CA_UPDATE) {
1478 // Tweak our audio processing settings, if needed.
1479 AudioOptions audio_options;
1480 if (!media_channel()->GetOptions(&audio_options)) {
1481 LOG(LS_WARNING) << "Can not set audio options from on remote content.";
1482 } else {
1483 if (audio->conference_mode()) {
1484 audio_options.conference_mode.Set(true);
1485 }
1486 if (audio->agc_minus_10db()) {
1487 audio_options.adjust_agc_delta.Set(kAgcMinus10db);
1488 }
1489 if (!media_channel()->SetOptions(audio_options)) {
1490 // Log an error on failure, but don't abort the call.
1491 LOG(LS_ERROR) << "Failed to set voice channel options";
1492 }
1493 }
1494 }
1495
1496 // If everything worked, see if we can start sending.
1497 if (ret) {
1498 ChangeState();
1499 } else {
1500 LOG(LS_WARNING) << "Failed to set remote voice description";
1501 }
1502 return ret;
1503}
1504
1505bool VoiceChannel::SetRingbackTone_w(const void* buf, int len) {
1506 ASSERT(worker_thread() == talk_base::Thread::Current());
1507 return media_channel()->SetRingbackTone(static_cast<const char*>(buf), len);
1508}
1509
1510bool VoiceChannel::PlayRingbackTone_w(uint32 ssrc, bool play, bool loop) {
1511 ASSERT(worker_thread() == talk_base::Thread::Current());
1512 if (play) {
1513 LOG(LS_INFO) << "Playing ringback tone, loop=" << loop;
1514 } else {
1515 LOG(LS_INFO) << "Stopping ringback tone";
1516 }
1517 return media_channel()->PlayRingbackTone(ssrc, play, loop);
1518}
1519
1520void VoiceChannel::HandleEarlyMediaTimeout() {
1521 // This occurs on the main thread, not the worker thread.
1522 if (!received_media_) {
1523 LOG(LS_INFO) << "No early media received before timeout";
1524 SignalEarlyMediaTimeout(this);
1525 }
1526}
1527
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001528bool VoiceChannel::InsertDtmf_w(uint32 ssrc, int event, int duration,
1529 int flags) {
1530 if (!enabled()) {
1531 return false;
1532 }
1533
1534 return media_channel()->InsertDtmf(ssrc, event, duration, flags);
1535}
1536
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001537bool VoiceChannel::SetChannelOptions(const AudioOptions& options) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001538 return InvokeOnWorker(Bind(&VoiceMediaChannel::SetOptions,
1539 media_channel(), options));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001540}
1541
1542void VoiceChannel::OnMessage(talk_base::Message *pmsg) {
1543 switch (pmsg->message_id) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001544 case MSG_EARLYMEDIATIMEOUT:
1545 HandleEarlyMediaTimeout();
1546 break;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001547 case MSG_CHANNEL_ERROR: {
1548 VoiceChannelErrorMessageData* data =
1549 static_cast<VoiceChannelErrorMessageData*>(pmsg->pdata);
1550 SignalMediaError(this, data->ssrc, data->error);
1551 delete data;
1552 break;
1553 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001554 default:
1555 BaseChannel::OnMessage(pmsg);
1556 break;
1557 }
1558}
1559
1560void VoiceChannel::OnConnectionMonitorUpdate(
1561 SocketMonitor* monitor, const std::vector<ConnectionInfo>& infos) {
1562 SignalConnectionMonitor(this, infos);
1563}
1564
1565void VoiceChannel::OnMediaMonitorUpdate(
1566 VoiceMediaChannel* media_channel, const VoiceMediaInfo& info) {
1567 ASSERT(media_channel == this->media_channel());
1568 SignalMediaMonitor(this, info);
1569}
1570
1571void VoiceChannel::OnAudioMonitorUpdate(AudioMonitor* monitor,
1572 const AudioInfo& info) {
1573 SignalAudioMonitor(this, info);
1574}
1575
1576void VoiceChannel::OnVoiceChannelError(
1577 uint32 ssrc, VoiceMediaChannel::Error err) {
1578 VoiceChannelErrorMessageData* data = new VoiceChannelErrorMessageData(
1579 ssrc, err);
1580 signaling_thread()->Post(this, MSG_CHANNEL_ERROR, data);
1581}
1582
1583void VoiceChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode,
1584 SrtpFilter::Error error) {
1585 switch (error) {
1586 case SrtpFilter::ERROR_FAIL:
1587 OnVoiceChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
1588 VoiceMediaChannel::ERROR_REC_SRTP_ERROR :
1589 VoiceMediaChannel::ERROR_PLAY_SRTP_ERROR);
1590 break;
1591 case SrtpFilter::ERROR_AUTH:
1592 OnVoiceChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
1593 VoiceMediaChannel::ERROR_REC_SRTP_AUTH_FAILED :
1594 VoiceMediaChannel::ERROR_PLAY_SRTP_AUTH_FAILED);
1595 break;
1596 case SrtpFilter::ERROR_REPLAY:
1597 // Only receving channel should have this error.
1598 ASSERT(mode == SrtpFilter::UNPROTECT);
1599 OnVoiceChannelError(ssrc, VoiceMediaChannel::ERROR_PLAY_SRTP_REPLAY);
1600 break;
1601 default:
1602 break;
1603 }
1604}
1605
1606void VoiceChannel::GetSrtpCiphers(std::vector<std::string>* ciphers) const {
1607 GetSupportedAudioCryptoSuites(ciphers);
1608}
1609
1610VideoChannel::VideoChannel(talk_base::Thread* thread,
1611 MediaEngineInterface* media_engine,
1612 VideoMediaChannel* media_channel,
1613 BaseSession* session,
1614 const std::string& content_name,
1615 bool rtcp,
1616 VoiceChannel* voice_channel)
1617 : BaseChannel(thread, media_engine, media_channel, session, content_name,
1618 rtcp),
1619 voice_channel_(voice_channel),
1620 renderer_(NULL),
1621 screencapture_factory_(CreateScreenCapturerFactory()),
1622 previous_we_(talk_base::WE_CLOSE) {
1623}
1624
1625bool VideoChannel::Init() {
1626 TransportChannel* rtcp_channel = rtcp() ? session()->CreateChannel(
1627 content_name(), "video_rtcp", ICE_CANDIDATE_COMPONENT_RTCP) : NULL;
1628 if (!BaseChannel::Init(session()->CreateChannel(
1629 content_name(), "video_rtp", ICE_CANDIDATE_COMPONENT_RTP),
1630 rtcp_channel)) {
1631 return false;
1632 }
1633 media_channel()->SignalMediaError.connect(
1634 this, &VideoChannel::OnVideoChannelError);
1635 srtp_filter()->SignalSrtpError.connect(
1636 this, &VideoChannel::OnSrtpError);
1637 return true;
1638}
1639
1640void VoiceChannel::SendLastMediaError() {
1641 uint32 ssrc;
1642 VoiceMediaChannel::Error error;
1643 media_channel()->GetLastMediaError(&ssrc, &error);
1644 SignalMediaError(this, ssrc, error);
1645}
1646
1647VideoChannel::~VideoChannel() {
1648 std::vector<uint32> screencast_ssrcs;
1649 ScreencastMap::iterator iter;
1650 while (!screencast_capturers_.empty()) {
1651 if (!RemoveScreencast(screencast_capturers_.begin()->first)) {
1652 LOG(LS_ERROR) << "Unable to delete screencast with ssrc "
1653 << screencast_capturers_.begin()->first;
1654 ASSERT(false);
1655 break;
1656 }
1657 }
1658
1659 StopMediaMonitor();
1660 // this can't be done in the base class, since it calls a virtual
1661 DisableMedia_w();
wu@webrtc.org78187522013-10-07 23:32:02 +00001662
1663 Deinit();
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001664}
1665
1666bool VideoChannel::SetRenderer(uint32 ssrc, VideoRenderer* renderer) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001667 worker_thread()->Invoke<void>(Bind(
1668 &VideoMediaChannel::SetRenderer, media_channel(), ssrc, renderer));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001669 return true;
1670}
1671
1672bool VideoChannel::ApplyViewRequest(const ViewRequest& request) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001673 return InvokeOnWorker(Bind(&VideoChannel::ApplyViewRequest_w, this, request));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001674}
1675
1676VideoCapturer* VideoChannel::AddScreencast(
1677 uint32 ssrc, const ScreencastId& id) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001678 return worker_thread()->Invoke<VideoCapturer*>(Bind(
1679 &VideoChannel::AddScreencast_w, this, ssrc, id));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001680}
1681
1682bool VideoChannel::SetCapturer(uint32 ssrc, VideoCapturer* capturer) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001683 return InvokeOnWorker(Bind(&VideoMediaChannel::SetCapturer,
1684 media_channel(), ssrc, capturer));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001685}
1686
1687bool VideoChannel::RemoveScreencast(uint32 ssrc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001688 return InvokeOnWorker(Bind(&VideoChannel::RemoveScreencast_w, this, ssrc));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001689}
1690
1691bool VideoChannel::IsScreencasting() {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001692 return InvokeOnWorker(Bind(&VideoChannel::IsScreencasting_w, this));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001693}
1694
wu@webrtc.orgcadf9042013-08-30 21:24:16 +00001695int VideoChannel::GetScreencastFps(uint32 ssrc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001696 ScreencastDetailsData data(ssrc);
1697 worker_thread()->Invoke<void>(Bind(
1698 &VideoChannel::GetScreencastDetails_w, this, &data));
wu@webrtc.orgcadf9042013-08-30 21:24:16 +00001699 return data.fps;
1700}
1701
1702int VideoChannel::GetScreencastMaxPixels(uint32 ssrc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001703 ScreencastDetailsData data(ssrc);
1704 worker_thread()->Invoke<void>(Bind(
1705 &VideoChannel::GetScreencastDetails_w, this, &data));
wu@webrtc.orgcadf9042013-08-30 21:24:16 +00001706 return data.screencast_max_pixels;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001707}
1708
1709bool VideoChannel::SendIntraFrame() {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001710 worker_thread()->Invoke<void>(Bind(
1711 &VideoMediaChannel::SendIntraFrame, media_channel()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001712 return true;
1713}
1714
1715bool VideoChannel::RequestIntraFrame() {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001716 worker_thread()->Invoke<void>(Bind(
1717 &VideoMediaChannel::RequestIntraFrame, media_channel()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001718 return true;
1719}
1720
1721void VideoChannel::SetScreenCaptureFactory(
1722 ScreenCapturerFactory* screencapture_factory) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001723 worker_thread()->Invoke<void>(Bind(
1724 &VideoChannel::SetScreenCaptureFactory_w,
1725 this, screencapture_factory));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001726}
1727
1728void VideoChannel::ChangeState() {
1729 // Render incoming data if we're the active call, and we have the local
1730 // content. We receive data on the default channel and multiplexed streams.
1731 bool recv = IsReadyToReceive();
1732 if (!media_channel()->SetRender(recv)) {
1733 LOG(LS_ERROR) << "Failed to SetRender on video channel";
1734 // TODO(gangji): Report error back to server.
1735 }
1736
1737 // Send outgoing data if we're the active call, we have the remote content,
1738 // and we have had some form of connectivity.
1739 bool send = IsReadyToSend();
1740 if (!media_channel()->SetSend(send)) {
1741 LOG(LS_ERROR) << "Failed to SetSend on video channel";
1742 // TODO(gangji): Report error back to server.
1743 }
1744
1745 LOG(LS_INFO) << "Changing video state, recv=" << recv << " send=" << send;
1746}
1747
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +00001748bool VideoChannel::GetStats(
1749 const StatsOptions& options, VideoMediaInfo* stats) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001750 return InvokeOnWorker(Bind(&VideoMediaChannel::GetStats,
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +00001751 media_channel(), options, stats));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001752}
1753
1754void VideoChannel::StartMediaMonitor(int cms) {
1755 media_monitor_.reset(new VideoMediaMonitor(media_channel(), worker_thread(),
1756 talk_base::Thread::Current()));
1757 media_monitor_->SignalUpdate.connect(
1758 this, &VideoChannel::OnMediaMonitorUpdate);
1759 media_monitor_->Start(cms);
1760}
1761
1762void VideoChannel::StopMediaMonitor() {
1763 if (media_monitor_) {
1764 media_monitor_->Stop();
1765 media_monitor_.reset();
1766 }
1767}
1768
1769const ContentInfo* VideoChannel::GetFirstContent(
1770 const SessionDescription* sdesc) {
1771 return GetFirstVideoContent(sdesc);
1772}
1773
1774bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001775 ContentAction action,
1776 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001777 ASSERT(worker_thread() == talk_base::Thread::Current());
1778 LOG(LS_INFO) << "Setting local video description";
1779
1780 const VideoContentDescription* video =
1781 static_cast<const VideoContentDescription*>(content);
1782 ASSERT(video != NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001783 if (!video) {
1784 SafeSetError("Can't find video content in local description.", error_desc);
1785 return false;
1786 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001787
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001788 bool ret = SetBaseLocalContent_w(content, action, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001789 // Set local video codecs (what we want to receive).
1790 if (action != CA_UPDATE || video->has_codecs()) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001791 if (!media_channel()->SetRecvCodecs(video->codecs())) {
1792 SafeSetError("Failed to set video receive codecs.", error_desc);
1793 ret = false;
1794 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001795 }
1796
1797 if (action != CA_UPDATE) {
1798 VideoOptions video_options;
1799 media_channel()->GetOptions(&video_options);
1800 video_options.buffered_mode_latency.Set(video->buffered_mode_latency());
1801
1802 if (!media_channel()->SetOptions(video_options)) {
1803 // Log an error on failure, but don't abort the call.
1804 LOG(LS_ERROR) << "Failed to set video channel options";
1805 }
1806 }
1807
1808 // If everything worked, see if we can start receiving.
1809 if (ret) {
1810 ChangeState();
1811 } else {
1812 LOG(LS_WARNING) << "Failed to set local video description";
1813 }
1814 return ret;
1815}
1816
1817bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001818 ContentAction action,
1819 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001820 ASSERT(worker_thread() == talk_base::Thread::Current());
1821 LOG(LS_INFO) << "Setting remote video description";
1822
1823 const VideoContentDescription* video =
1824 static_cast<const VideoContentDescription*>(content);
1825 ASSERT(video != NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001826 if (!video) {
1827 SafeSetError("Can't find video content in remote description.", error_desc);
1828 return false;
1829 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001830
1831 bool ret = true;
1832 // Set remote video codecs (what the other side wants to receive).
1833 if (action != CA_UPDATE || video->has_codecs()) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001834 if (!media_channel()->SetSendCodecs(video->codecs())) {
1835 SafeSetError("Failed to set video send codecs.", error_desc);
1836 ret = false;
1837 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001838 }
1839
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001840 ret &= SetBaseRemoteContent_w(content, action, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001841
1842 if (action != CA_UPDATE) {
1843 // Tweak our video processing settings, if needed.
1844 VideoOptions video_options;
1845 media_channel()->GetOptions(&video_options);
wu@webrtc.org9caf2762013-12-11 18:25:07 +00001846 if (video->conference_mode()) {
1847 video_options.conference_mode.Set(true);
1848 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001849 video_options.buffered_mode_latency.Set(video->buffered_mode_latency());
1850
1851 if (!media_channel()->SetOptions(video_options)) {
1852 // Log an error on failure, but don't abort the call.
1853 LOG(LS_ERROR) << "Failed to set video channel options";
1854 }
1855 }
1856
1857 // If everything worked, see if we can start sending.
1858 if (ret) {
1859 ChangeState();
1860 } else {
1861 LOG(LS_WARNING) << "Failed to set remote video description";
1862 }
1863 return ret;
1864}
1865
1866bool VideoChannel::ApplyViewRequest_w(const ViewRequest& request) {
1867 bool ret = true;
1868 // Set the send format for each of the local streams. If the view request
1869 // does not contain a local stream, set its send format to 0x0, which will
1870 // drop all frames.
1871 for (std::vector<StreamParams>::const_iterator it = local_streams().begin();
1872 it != local_streams().end(); ++it) {
1873 VideoFormat format(0, 0, 0, cricket::FOURCC_I420);
1874 StaticVideoViews::const_iterator view;
1875 for (view = request.static_video_views.begin();
1876 view != request.static_video_views.end(); ++view) {
1877 if (view->selector.Matches(*it)) {
1878 format.width = view->width;
1879 format.height = view->height;
1880 format.interval = cricket::VideoFormat::FpsToInterval(view->framerate);
1881 break;
1882 }
1883 }
1884
1885 ret &= media_channel()->SetSendStreamFormat(it->first_ssrc(), format);
1886 }
1887
1888 // Check if the view request has invalid streams.
1889 for (StaticVideoViews::const_iterator it = request.static_video_views.begin();
1890 it != request.static_video_views.end(); ++it) {
1891 if (!GetStream(local_streams(), it->selector, NULL)) {
1892 LOG(LS_WARNING) << "View request for ("
1893 << it->selector.ssrc << ", '"
1894 << it->selector.groupid << "', '"
1895 << it->selector.streamid << "'"
1896 << ") is not in the local streams.";
1897 }
1898 }
1899
1900 return ret;
1901}
1902
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001903VideoCapturer* VideoChannel::AddScreencast_w(
1904 uint32 ssrc, const ScreencastId& id) {
1905 if (screencast_capturers_.find(ssrc) != screencast_capturers_.end()) {
1906 return NULL;
1907 }
1908 VideoCapturer* screen_capturer =
1909 screencapture_factory_->CreateScreenCapturer(id);
1910 if (!screen_capturer) {
1911 return NULL;
1912 }
1913 screen_capturer->SignalStateChange.connect(this,
1914 &VideoChannel::OnStateChange);
1915 screencast_capturers_[ssrc] = screen_capturer;
1916 return screen_capturer;
1917}
1918
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001919bool VideoChannel::RemoveScreencast_w(uint32 ssrc) {
1920 ScreencastMap::iterator iter = screencast_capturers_.find(ssrc);
1921 if (iter == screencast_capturers_.end()) {
1922 return false;
1923 }
1924 // Clean up VideoCapturer.
1925 delete iter->second;
1926 screencast_capturers_.erase(iter);
1927 return true;
1928}
1929
1930bool VideoChannel::IsScreencasting_w() const {
1931 return !screencast_capturers_.empty();
1932}
1933
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001934void VideoChannel::GetScreencastDetails_w(
1935 ScreencastDetailsData* data) const {
wu@webrtc.orgcadf9042013-08-30 21:24:16 +00001936 ScreencastMap::const_iterator iter = screencast_capturers_.find(data->ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001937 if (iter == screencast_capturers_.end()) {
wu@webrtc.orgcadf9042013-08-30 21:24:16 +00001938 return;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001939 }
1940 VideoCapturer* capturer = iter->second;
1941 const VideoFormat* video_format = capturer->GetCaptureFormat();
wu@webrtc.orgcadf9042013-08-30 21:24:16 +00001942 data->fps = VideoFormat::IntervalToFps(video_format->interval);
1943 data->screencast_max_pixels = capturer->screencast_max_pixels();
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001944}
1945
1946void VideoChannel::SetScreenCaptureFactory_w(
1947 ScreenCapturerFactory* screencapture_factory) {
1948 if (screencapture_factory == NULL) {
1949 screencapture_factory_.reset(CreateScreenCapturerFactory());
1950 } else {
1951 screencapture_factory_.reset(screencapture_factory);
1952 }
1953}
1954
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001955void VideoChannel::OnScreencastWindowEvent_s(uint32 ssrc,
1956 talk_base::WindowEvent we) {
1957 ASSERT(signaling_thread() == talk_base::Thread::Current());
1958 SignalScreencastWindowEvent(ssrc, we);
1959}
1960
1961bool VideoChannel::SetChannelOptions(const VideoOptions &options) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001962 return InvokeOnWorker(Bind(&VideoMediaChannel::SetOptions,
1963 media_channel(), options));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001964}
1965
1966void VideoChannel::OnMessage(talk_base::Message *pmsg) {
1967 switch (pmsg->message_id) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001968 case MSG_SCREENCASTWINDOWEVENT: {
1969 const ScreencastEventMessageData* data =
1970 static_cast<ScreencastEventMessageData*>(pmsg->pdata);
1971 OnScreencastWindowEvent_s(data->ssrc, data->event);
1972 delete data;
1973 break;
1974 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001975 case MSG_CHANNEL_ERROR: {
1976 const VideoChannelErrorMessageData* data =
1977 static_cast<VideoChannelErrorMessageData*>(pmsg->pdata);
1978 SignalMediaError(this, data->ssrc, data->error);
1979 delete data;
1980 break;
1981 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001982 default:
1983 BaseChannel::OnMessage(pmsg);
1984 break;
1985 }
1986}
1987
1988void VideoChannel::OnConnectionMonitorUpdate(
1989 SocketMonitor *monitor, const std::vector<ConnectionInfo> &infos) {
1990 SignalConnectionMonitor(this, infos);
1991}
1992
1993// TODO(pthatcher): Look into removing duplicate code between
1994// audio, video, and data, perhaps by using templates.
1995void VideoChannel::OnMediaMonitorUpdate(
1996 VideoMediaChannel* media_channel, const VideoMediaInfo &info) {
1997 ASSERT(media_channel == this->media_channel());
1998 SignalMediaMonitor(this, info);
1999}
2000
2001void VideoChannel::OnScreencastWindowEvent(uint32 ssrc,
2002 talk_base::WindowEvent event) {
2003 ScreencastEventMessageData* pdata =
2004 new ScreencastEventMessageData(ssrc, event);
2005 signaling_thread()->Post(this, MSG_SCREENCASTWINDOWEVENT, pdata);
2006}
2007
2008void VideoChannel::OnStateChange(VideoCapturer* capturer, CaptureState ev) {
2009 // Map capturer events to window events. In the future we may want to simply
2010 // pass these events up directly.
2011 talk_base::WindowEvent we;
2012 if (ev == CS_STOPPED) {
2013 we = talk_base::WE_CLOSE;
2014 } else if (ev == CS_PAUSED) {
2015 we = talk_base::WE_MINIMIZE;
2016 } else if (ev == CS_RUNNING && previous_we_ == talk_base::WE_MINIMIZE) {
2017 we = talk_base::WE_RESTORE;
2018 } else {
2019 return;
2020 }
2021 previous_we_ = we;
2022
2023 uint32 ssrc = 0;
2024 if (!GetLocalSsrc(capturer, &ssrc)) {
2025 return;
2026 }
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00002027
2028 OnScreencastWindowEvent(ssrc, we);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002029}
2030
2031bool VideoChannel::GetLocalSsrc(const VideoCapturer* capturer, uint32* ssrc) {
2032 *ssrc = 0;
2033 for (ScreencastMap::iterator iter = screencast_capturers_.begin();
2034 iter != screencast_capturers_.end(); ++iter) {
2035 if (iter->second == capturer) {
2036 *ssrc = iter->first;
2037 return true;
2038 }
2039 }
2040 return false;
2041}
2042
2043void VideoChannel::OnVideoChannelError(uint32 ssrc,
2044 VideoMediaChannel::Error error) {
2045 VideoChannelErrorMessageData* data = new VideoChannelErrorMessageData(
2046 ssrc, error);
2047 signaling_thread()->Post(this, MSG_CHANNEL_ERROR, data);
2048}
2049
2050void VideoChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode,
2051 SrtpFilter::Error error) {
2052 switch (error) {
2053 case SrtpFilter::ERROR_FAIL:
2054 OnVideoChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
2055 VideoMediaChannel::ERROR_REC_SRTP_ERROR :
2056 VideoMediaChannel::ERROR_PLAY_SRTP_ERROR);
2057 break;
2058 case SrtpFilter::ERROR_AUTH:
2059 OnVideoChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
2060 VideoMediaChannel::ERROR_REC_SRTP_AUTH_FAILED :
2061 VideoMediaChannel::ERROR_PLAY_SRTP_AUTH_FAILED);
2062 break;
2063 case SrtpFilter::ERROR_REPLAY:
2064 // Only receving channel should have this error.
2065 ASSERT(mode == SrtpFilter::UNPROTECT);
2066 // TODO(gangji): Turn on the signaling of replay error once we have
2067 // switched to the new mechanism for doing video retransmissions.
2068 // OnVideoChannelError(ssrc, VideoMediaChannel::ERROR_PLAY_SRTP_REPLAY);
2069 break;
2070 default:
2071 break;
2072 }
2073}
2074
2075
2076void VideoChannel::GetSrtpCiphers(std::vector<std::string>* ciphers) const {
2077 GetSupportedVideoCryptoSuites(ciphers);
2078}
2079
2080DataChannel::DataChannel(talk_base::Thread* thread,
2081 DataMediaChannel* media_channel,
2082 BaseSession* session,
2083 const std::string& content_name,
2084 bool rtcp)
2085 // MediaEngine is NULL
2086 : BaseChannel(thread, NULL, media_channel, session, content_name, rtcp),
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +00002087 data_channel_type_(cricket::DCT_NONE),
2088 ready_to_send_data_(false) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002089}
2090
2091DataChannel::~DataChannel() {
2092 StopMediaMonitor();
2093 // this can't be done in the base class, since it calls a virtual
2094 DisableMedia_w();
wu@webrtc.org78187522013-10-07 23:32:02 +00002095
2096 Deinit();
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002097}
2098
2099bool DataChannel::Init() {
2100 TransportChannel* rtcp_channel = rtcp() ? session()->CreateChannel(
2101 content_name(), "data_rtcp", ICE_CANDIDATE_COMPONENT_RTCP) : NULL;
2102 if (!BaseChannel::Init(session()->CreateChannel(
2103 content_name(), "data_rtp", ICE_CANDIDATE_COMPONENT_RTP),
2104 rtcp_channel)) {
2105 return false;
2106 }
2107 media_channel()->SignalDataReceived.connect(
2108 this, &DataChannel::OnDataReceived);
2109 media_channel()->SignalMediaError.connect(
2110 this, &DataChannel::OnDataChannelError);
wu@webrtc.orgd64719d2013-08-01 00:00:07 +00002111 media_channel()->SignalReadyToSend.connect(
2112 this, &DataChannel::OnDataChannelReadyToSend);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002113 srtp_filter()->SignalSrtpError.connect(
2114 this, &DataChannel::OnSrtpError);
2115 return true;
2116}
2117
2118bool DataChannel::SendData(const SendDataParams& params,
2119 const talk_base::Buffer& payload,
2120 SendDataResult* result) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00002121 return InvokeOnWorker(Bind(&DataMediaChannel::SendData,
2122 media_channel(), params, payload, result));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002123}
2124
2125const ContentInfo* DataChannel::GetFirstContent(
2126 const SessionDescription* sdesc) {
2127 return GetFirstDataContent(sdesc);
2128}
2129
2130
2131static bool IsRtpPacket(const talk_base::Buffer* packet) {
2132 int version;
2133 if (!GetRtpVersion(packet->data(), packet->length(), &version)) {
2134 return false;
2135 }
2136
2137 return version == 2;
2138}
2139
2140bool DataChannel::WantsPacket(bool rtcp, talk_base::Buffer* packet) {
2141 if (data_channel_type_ == DCT_SCTP) {
2142 // TODO(pthatcher): Do this in a more robust way by checking for
2143 // SCTP or DTLS.
2144 return !IsRtpPacket(packet);
2145 } else if (data_channel_type_ == DCT_RTP) {
2146 return BaseChannel::WantsPacket(rtcp, packet);
2147 }
2148 return false;
2149}
2150
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002151bool DataChannel::SetDataChannelType(DataChannelType new_data_channel_type,
2152 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002153 // It hasn't been set before, so set it now.
2154 if (data_channel_type_ == DCT_NONE) {
2155 data_channel_type_ = new_data_channel_type;
2156 return true;
2157 }
2158
2159 // It's been set before, but doesn't match. That's bad.
2160 if (data_channel_type_ != new_data_channel_type) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002161 std::ostringstream desc;
2162 desc << "Data channel type mismatch."
2163 << " Expected " << data_channel_type_
2164 << " Got " << new_data_channel_type;
2165 SafeSetError(desc.str(), error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002166 return false;
2167 }
2168
2169 // It's hasn't changed. Nothing to do.
2170 return true;
2171}
2172
2173bool DataChannel::SetDataChannelTypeFromContent(
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002174 const DataContentDescription* content,
2175 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002176 bool is_sctp = ((content->protocol() == kMediaProtocolSctp) ||
2177 (content->protocol() == kMediaProtocolDtlsSctp));
2178 DataChannelType data_channel_type = is_sctp ? DCT_SCTP : DCT_RTP;
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002179 return SetDataChannelType(data_channel_type, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002180}
2181
2182bool DataChannel::SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002183 ContentAction action,
2184 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002185 ASSERT(worker_thread() == talk_base::Thread::Current());
2186 LOG(LS_INFO) << "Setting local data description";
2187
2188 const DataContentDescription* data =
2189 static_cast<const DataContentDescription*>(content);
2190 ASSERT(data != NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002191 if (!data) {
2192 SafeSetError("Can't find data content in local description.", error_desc);
2193 return false;
2194 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002195
2196 bool ret = false;
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002197 if (!SetDataChannelTypeFromContent(data, error_desc)) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002198 return false;
2199 }
2200
2201 if (data_channel_type_ == DCT_SCTP) {
2202 // SCTP data channels don't need the rest of the stuff.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002203 ret = UpdateLocalStreams_w(data->streams(), action, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002204 if (ret) {
2205 set_local_content_direction(content->direction());
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00002206 // As in SetRemoteContent_w, make sure we set the local SCTP port
2207 // number as specified in our DataContentDescription.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002208 if (!media_channel()->SetRecvCodecs(data->codecs())) {
2209 SafeSetError("Failed to set data receive codecs.", error_desc);
2210 ret = false;
2211 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002212 }
2213 } else {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002214 ret = SetBaseLocalContent_w(content, action, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002215
2216 if (action != CA_UPDATE || data->has_codecs()) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002217 if (!media_channel()->SetRecvCodecs(data->codecs())) {
2218 SafeSetError("Failed to set data receive codecs.", error_desc);
2219 ret = false;
2220 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002221 }
2222 }
2223
2224 // If everything worked, see if we can start receiving.
2225 if (ret) {
2226 ChangeState();
2227 } else {
2228 LOG(LS_WARNING) << "Failed to set local data description";
2229 }
2230 return ret;
2231}
2232
2233bool DataChannel::SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002234 ContentAction action,
2235 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002236 ASSERT(worker_thread() == talk_base::Thread::Current());
2237
2238 const DataContentDescription* data =
2239 static_cast<const DataContentDescription*>(content);
2240 ASSERT(data != NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002241 if (!data) {
2242 SafeSetError("Can't find data content in remote description.", error_desc);
2243 return false;
2244 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002245
2246 bool ret = true;
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002247 if (!SetDataChannelTypeFromContent(data, error_desc)) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002248 return false;
2249 }
2250
2251 if (data_channel_type_ == DCT_SCTP) {
2252 LOG(LS_INFO) << "Setting SCTP remote data description";
2253 // SCTP data channels don't need the rest of the stuff.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002254 ret = UpdateRemoteStreams_w(content->streams(), action, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002255 if (ret) {
2256 set_remote_content_direction(content->direction());
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00002257 // We send the SCTP port number (not to be confused with the underlying
2258 // UDP port number) as a codec parameter. Make sure it gets there.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002259 if (!media_channel()->SetSendCodecs(data->codecs())) {
2260 SafeSetError("Failed to set data send codecs.", error_desc);
2261 ret = false;
2262 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002263 }
2264 } else {
2265 // If the remote data doesn't have codecs and isn't an update, it
2266 // must be empty, so ignore it.
2267 if (action != CA_UPDATE && !data->has_codecs()) {
2268 return true;
2269 }
2270 LOG(LS_INFO) << "Setting remote data description";
2271
2272 // Set remote video codecs (what the other side wants to receive).
2273 if (action != CA_UPDATE || data->has_codecs()) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002274 if (!media_channel()->SetSendCodecs(data->codecs())) {
2275 SafeSetError("Failed to set data send codecs.", error_desc);
2276 ret = false;
2277 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002278 }
2279
2280 if (ret) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002281 ret &= SetBaseRemoteContent_w(content, action, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002282 }
2283
2284 if (action != CA_UPDATE) {
2285 int bandwidth_bps = data->bandwidth();
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002286 if (!media_channel()->SetMaxSendBandwidth(bandwidth_bps)) {
2287 std::ostringstream desc;
2288 desc << "Failed to set max send bandwidth for data content.";
2289 SafeSetError(desc.str(), error_desc);
2290 ret = false;
2291 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002292 }
2293 }
2294
2295 // If everything worked, see if we can start sending.
2296 if (ret) {
2297 ChangeState();
2298 } else {
2299 LOG(LS_WARNING) << "Failed to set remote data description";
2300 }
2301 return ret;
2302}
2303
2304void DataChannel::ChangeState() {
2305 // Render incoming data if we're the active call, and we have the local
2306 // content. We receive data on the default channel and multiplexed streams.
2307 bool recv = IsReadyToReceive();
2308 if (!media_channel()->SetReceive(recv)) {
2309 LOG(LS_ERROR) << "Failed to SetReceive on data channel";
2310 }
2311
2312 // Send outgoing data if we're the active call, we have the remote content,
2313 // and we have had some form of connectivity.
2314 bool send = IsReadyToSend();
2315 if (!media_channel()->SetSend(send)) {
2316 LOG(LS_ERROR) << "Failed to SetSend on data channel";
2317 }
2318
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00002319 // Trigger SignalReadyToSendData asynchronously.
2320 OnDataChannelReadyToSend(send);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002321
2322 LOG(LS_INFO) << "Changing data state, recv=" << recv << " send=" << send;
2323}
2324
2325void DataChannel::OnMessage(talk_base::Message *pmsg) {
2326 switch (pmsg->message_id) {
2327 case MSG_READYTOSENDDATA: {
wu@webrtc.orgd64719d2013-08-01 00:00:07 +00002328 DataChannelReadyToSendMessageData* data =
2329 static_cast<DataChannelReadyToSendMessageData*>(pmsg->pdata);
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +00002330 ready_to_send_data_ = data->data();
2331 SignalReadyToSendData(ready_to_send_data_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002332 delete data;
2333 break;
2334 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002335 case MSG_DATARECEIVED: {
2336 DataReceivedMessageData* data =
2337 static_cast<DataReceivedMessageData*>(pmsg->pdata);
2338 SignalDataReceived(this, data->params, data->payload);
2339 delete data;
2340 break;
2341 }
2342 case MSG_CHANNEL_ERROR: {
2343 const DataChannelErrorMessageData* data =
2344 static_cast<DataChannelErrorMessageData*>(pmsg->pdata);
2345 SignalMediaError(this, data->ssrc, data->error);
2346 delete data;
2347 break;
2348 }
2349 default:
2350 BaseChannel::OnMessage(pmsg);
2351 break;
2352 }
2353}
2354
2355void DataChannel::OnConnectionMonitorUpdate(
2356 SocketMonitor* monitor, const std::vector<ConnectionInfo>& infos) {
2357 SignalConnectionMonitor(this, infos);
2358}
2359
2360void DataChannel::StartMediaMonitor(int cms) {
2361 media_monitor_.reset(new DataMediaMonitor(media_channel(), worker_thread(),
2362 talk_base::Thread::Current()));
2363 media_monitor_->SignalUpdate.connect(
2364 this, &DataChannel::OnMediaMonitorUpdate);
2365 media_monitor_->Start(cms);
2366}
2367
2368void DataChannel::StopMediaMonitor() {
2369 if (media_monitor_) {
2370 media_monitor_->Stop();
2371 media_monitor_->SignalUpdate.disconnect(this);
2372 media_monitor_.reset();
2373 }
2374}
2375
2376void DataChannel::OnMediaMonitorUpdate(
2377 DataMediaChannel* media_channel, const DataMediaInfo& info) {
2378 ASSERT(media_channel == this->media_channel());
2379 SignalMediaMonitor(this, info);
2380}
2381
2382void DataChannel::OnDataReceived(
2383 const ReceiveDataParams& params, const char* data, size_t len) {
2384 DataReceivedMessageData* msg = new DataReceivedMessageData(
2385 params, data, len);
2386 signaling_thread()->Post(this, MSG_DATARECEIVED, msg);
2387}
2388
2389void DataChannel::OnDataChannelError(
2390 uint32 ssrc, DataMediaChannel::Error err) {
2391 DataChannelErrorMessageData* data = new DataChannelErrorMessageData(
2392 ssrc, err);
2393 signaling_thread()->Post(this, MSG_CHANNEL_ERROR, data);
2394}
2395
wu@webrtc.orgd64719d2013-08-01 00:00:07 +00002396void DataChannel::OnDataChannelReadyToSend(bool writable) {
2397 // This is usded for congestion control to indicate that the stream is ready
2398 // to send by the MediaChannel, as opposed to OnReadyToSend, which indicates
2399 // that the transport channel is ready.
2400 signaling_thread()->Post(this, MSG_READYTOSENDDATA,
2401 new DataChannelReadyToSendMessageData(writable));
2402}
2403
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002404void DataChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode,
2405 SrtpFilter::Error error) {
2406 switch (error) {
2407 case SrtpFilter::ERROR_FAIL:
2408 OnDataChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
2409 DataMediaChannel::ERROR_SEND_SRTP_ERROR :
2410 DataMediaChannel::ERROR_RECV_SRTP_ERROR);
2411 break;
2412 case SrtpFilter::ERROR_AUTH:
2413 OnDataChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
2414 DataMediaChannel::ERROR_SEND_SRTP_AUTH_FAILED :
2415 DataMediaChannel::ERROR_RECV_SRTP_AUTH_FAILED);
2416 break;
2417 case SrtpFilter::ERROR_REPLAY:
2418 // Only receving channel should have this error.
2419 ASSERT(mode == SrtpFilter::UNPROTECT);
2420 OnDataChannelError(ssrc, DataMediaChannel::ERROR_RECV_SRTP_REPLAY);
2421 break;
2422 default:
2423 break;
2424 }
2425}
2426
2427void DataChannel::GetSrtpCiphers(std::vector<std::string>* ciphers) const {
2428 GetSupportedDataCryptoSuites(ciphers);
2429}
2430
2431bool DataChannel::ShouldSetupDtlsSrtp() const {
2432 return (data_channel_type_ == DCT_RTP);
2433}
2434
2435} // namespace cricket