blob: a25ab7d0df682791ec5b4abc0d07f4efae3166df [file] [log] [blame]
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
buildbot@webrtc.org5b1ebac2014-08-07 17:18:00 +000030#include "talk/media/base/constants.h"
31#include "talk/media/base/rtputils.h"
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000032#include "webrtc/p2p/base/transportchannel.h"
buildbot@webrtc.org5b1ebac2014-08-07 17:18:00 +000033#include "talk/session/media/channelmanager.h"
buildbot@webrtc.org65b98d12014-08-07 22:09:08 +000034#include "webrtc/base/bind.h"
35#include "webrtc/base/buffer.h"
36#include "webrtc/base/byteorder.h"
37#include "webrtc/base/common.h"
38#include "webrtc/base/dscp.h"
39#include "webrtc/base/logging.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000040
41namespace cricket {
42
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000043using rtc::Bind;
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +000044
henrike@webrtc.org28e20752013-07-10 00:45:36 +000045enum {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +000046 MSG_EARLYMEDIATIMEOUT = 1,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000047 MSG_SCREENCASTWINDOWEVENT,
48 MSG_RTPPACKET,
49 MSG_RTCPPACKET,
50 MSG_CHANNEL_ERROR,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000051 MSG_READYTOSENDDATA,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000052 MSG_DATARECEIVED,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000053 MSG_FIRSTPACKETRECEIVED,
buildbot@webrtc.org1d66be22014-05-29 22:54:24 +000054 MSG_STREAMCLOSEDREMOTELY,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000055};
56
57// Value specified in RFC 5764.
58static const char kDtlsSrtpExporterLabel[] = "EXTRACTOR-dtls_srtp";
59
60static const int kAgcMinus10db = -10;
61
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000062static void SafeSetError(const std::string& message, std::string* error_desc) {
63 if (error_desc) {
64 *error_desc = message;
65 }
66}
67
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000068struct PacketMessageData : public rtc::MessageData {
69 rtc::Buffer packet;
stefanc1aeaf02015-10-15 07:26:07 -070070 rtc::PacketOptions options;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000071};
72
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000073struct ScreencastEventMessageData : public rtc::MessageData {
Peter Boström0c4e06b2015-10-07 12:23:21 +020074 ScreencastEventMessageData(uint32_t s, rtc::WindowEvent we)
75 : ssrc(s), event(we) {}
76 uint32_t ssrc;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000077 rtc::WindowEvent event;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000078};
79
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000080struct VoiceChannelErrorMessageData : public rtc::MessageData {
Peter Boström0c4e06b2015-10-07 12:23:21 +020081 VoiceChannelErrorMessageData(uint32_t in_ssrc,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000082 VoiceMediaChannel::Error in_error)
Peter Boström0c4e06b2015-10-07 12:23:21 +020083 : ssrc(in_ssrc), error(in_error) {}
84 uint32_t ssrc;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000085 VoiceMediaChannel::Error error;
86};
87
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000088struct VideoChannelErrorMessageData : public rtc::MessageData {
Peter Boström0c4e06b2015-10-07 12:23:21 +020089 VideoChannelErrorMessageData(uint32_t in_ssrc,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000090 VideoMediaChannel::Error in_error)
Peter Boström0c4e06b2015-10-07 12:23:21 +020091 : ssrc(in_ssrc), error(in_error) {}
92 uint32_t ssrc;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000093 VideoMediaChannel::Error error;
94};
95
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000096struct DataChannelErrorMessageData : public rtc::MessageData {
Peter Boström0c4e06b2015-10-07 12:23:21 +020097 DataChannelErrorMessageData(uint32_t in_ssrc,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000098 DataMediaChannel::Error in_error)
Peter Boström0c4e06b2015-10-07 12:23:21 +020099 : ssrc(in_ssrc), error(in_error) {}
100 uint32_t ssrc;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000101 DataMediaChannel::Error error;
102};
103
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000104
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000105struct VideoChannel::ScreencastDetailsData {
Peter Boström0c4e06b2015-10-07 12:23:21 +0200106 explicit ScreencastDetailsData(uint32_t s)
107 : ssrc(s), fps(0), screencast_max_pixels(0) {}
108 uint32_t ssrc;
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000109 int fps;
110 int screencast_max_pixels;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000111};
112
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000113static const char* PacketType(bool rtcp) {
114 return (!rtcp) ? "RTP" : "RTCP";
115}
116
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000117static bool ValidPacket(bool rtcp, const rtc::Buffer* packet) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000118 // Check the packet size. We could check the header too if needed.
119 return (packet &&
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000120 packet->size() >= (!rtcp ? kMinRtpPacketLen : kMinRtcpPacketLen) &&
121 packet->size() <= kMaxRtpPacketLen);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000122}
123
124static bool IsReceiveContentDirection(MediaContentDirection direction) {
125 return direction == MD_SENDRECV || direction == MD_RECVONLY;
126}
127
128static bool IsSendContentDirection(MediaContentDirection direction) {
129 return direction == MD_SENDRECV || direction == MD_SENDONLY;
130}
131
132static const MediaContentDescription* GetContentDescription(
133 const ContentInfo* cinfo) {
134 if (cinfo == NULL)
135 return NULL;
136 return static_cast<const MediaContentDescription*>(cinfo->description);
137}
138
Peter Thatcherc2ee2c82015-08-07 16:05:34 -0700139template <class Codec>
140void RtpParametersFromMediaDescription(
141 const MediaContentDescriptionImpl<Codec>* desc,
142 RtpParameters<Codec>* params) {
143 // TODO(pthatcher): Remove this once we're sure no one will give us
144 // a description without codecs (currently a CA_UPDATE with just
145 // streams can).
146 if (desc->has_codecs()) {
147 params->codecs = desc->codecs();
148 }
149 // TODO(pthatcher): See if we really need
150 // rtp_header_extensions_set() and remove it if we don't.
151 if (desc->rtp_header_extensions_set()) {
152 params->extensions = desc->rtp_header_extensions();
153 }
154}
155
156template <class Codec, class Options>
157void RtpSendParametersFromMediaDescription(
158 const MediaContentDescriptionImpl<Codec>* desc,
159 RtpSendParameters<Codec, Options>* send_params) {
160 RtpParametersFromMediaDescription(desc, send_params);
161 send_params->max_bandwidth_bps = desc->bandwidth();
162}
163
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000164BaseChannel::BaseChannel(rtc::Thread* thread,
deadbeefcbecd352015-09-23 11:50:27 -0700165 MediaChannel* media_channel,
166 TransportController* transport_controller,
167 const std::string& content_name,
168 bool rtcp)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000169 : worker_thread_(thread),
deadbeefcbecd352015-09-23 11:50:27 -0700170 transport_controller_(transport_controller),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000171 media_channel_(media_channel),
172 content_name_(content_name),
deadbeefcbecd352015-09-23 11:50:27 -0700173 rtcp_transport_enabled_(rtcp),
174 transport_channel_(nullptr),
175 rtcp_transport_channel_(nullptr),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000176 enabled_(false),
177 writable_(false),
178 rtp_ready_to_send_(false),
179 rtcp_ready_to_send_(false),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000180 was_ever_writable_(false),
181 local_content_direction_(MD_INACTIVE),
182 remote_content_direction_(MD_INACTIVE),
183 has_received_packet_(false),
184 dtls_keyed_(false),
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000185 secure_required_(false),
186 rtp_abs_sendtime_extn_id_(-1) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000187 ASSERT(worker_thread_ == rtc::Thread::Current());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000188 LOG(LS_INFO) << "Created channel for " << content_name;
189}
190
191BaseChannel::~BaseChannel() {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000192 ASSERT(worker_thread_ == rtc::Thread::Current());
wu@webrtc.org78187522013-10-07 23:32:02 +0000193 Deinit();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000194 StopConnectionMonitor();
195 FlushRtcpMessages(); // Send any outstanding RTCP packets.
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000196 worker_thread_->Clear(this); // eats any outstanding messages or packets
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000197 // We must destroy the media channel before the transport channel, otherwise
198 // the media channel may try to send on the dead transport channel. NULLing
199 // is not an effective strategy since the sends will come on another thread.
200 delete media_channel_;
deadbeefcbecd352015-09-23 11:50:27 -0700201 // Note that we don't just call set_transport_channel(nullptr) because that
202 // would call a pure virtual method which we can't do from a destructor.
203 if (transport_channel_) {
204 DisconnectFromTransportChannel(transport_channel_);
205 transport_controller_->DestroyTransportChannel_w(
206 transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTP);
207 }
208 if (rtcp_transport_channel_) {
209 DisconnectFromTransportChannel(rtcp_transport_channel_);
210 transport_controller_->DestroyTransportChannel_w(
211 transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
212 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000213 LOG(LS_INFO) << "Destroyed channel";
214}
215
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000216bool BaseChannel::Init() {
deadbeefcbecd352015-09-23 11:50:27 -0700217 if (!SetTransport(content_name())) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000218 return false;
219 }
220
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800221 if (!SetDtlsSrtpCryptoSuites(transport_channel(), false)) {
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000222 return false;
223 }
deadbeefcbecd352015-09-23 11:50:27 -0700224 if (rtcp_transport_enabled() &&
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800225 !SetDtlsSrtpCryptoSuites(rtcp_transport_channel(), true)) {
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000226 return false;
227 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000228
wu@webrtc.orgde305012013-10-31 15:40:38 +0000229 // Both RTP and RTCP channels are set, we can call SetInterface on
230 // media channel and it can set network options.
231 media_channel_->SetInterface(this);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000232 return true;
233}
234
wu@webrtc.org78187522013-10-07 23:32:02 +0000235void BaseChannel::Deinit() {
236 media_channel_->SetInterface(NULL);
237}
238
deadbeefcbecd352015-09-23 11:50:27 -0700239bool BaseChannel::SetTransport(const std::string& transport_name) {
240 return worker_thread_->Invoke<bool>(
241 Bind(&BaseChannel::SetTransport_w, this, transport_name));
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000242}
243
deadbeefcbecd352015-09-23 11:50:27 -0700244bool BaseChannel::SetTransport_w(const std::string& transport_name) {
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000245 ASSERT(worker_thread_ == rtc::Thread::Current());
246
deadbeefcbecd352015-09-23 11:50:27 -0700247 if (transport_name == transport_name_) {
248 // Nothing to do if transport name isn't changing
249 return true;
250 }
251
Guo-wei Shieh9c38c2d2015-12-05 09:46:07 -0800252 // When using DTLS-SRTP, we must reset the SrtpFilter every time the transport
253 // changes and wait until the DTLS handshake is complete to set the newly
254 // negotiated parameters.
255 if (ShouldSetupDtlsSrtp()) {
256 srtp_filter_.ResetParams();
257 }
258
deadbeefcbecd352015-09-23 11:50:27 -0700259 set_transport_channel(transport_controller_->CreateTransportChannel_w(
260 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP));
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000261 if (!transport_channel()) {
262 return false;
263 }
deadbeefcbecd352015-09-23 11:50:27 -0700264 if (rtcp_transport_enabled()) {
265 LOG(LS_INFO) << "Create RTCP TransportChannel for " << content_name()
266 << " on " << transport_name << " transport ";
267 set_rtcp_transport_channel(transport_controller_->CreateTransportChannel_w(
268 transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP));
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000269 if (!rtcp_transport_channel()) {
270 return false;
271 }
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000272 }
273
deadbeefcbecd352015-09-23 11:50:27 -0700274 transport_name_ = transport_name;
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000275 return true;
276}
277
278void BaseChannel::set_transport_channel(TransportChannel* new_tc) {
279 ASSERT(worker_thread_ == rtc::Thread::Current());
280
281 TransportChannel* old_tc = transport_channel_;
deadbeefcbecd352015-09-23 11:50:27 -0700282 if (!old_tc && !new_tc) {
283 // Nothing to do
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000284 return;
285 }
deadbeefcbecd352015-09-23 11:50:27 -0700286 ASSERT(old_tc != new_tc);
287
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000288 if (old_tc) {
289 DisconnectFromTransportChannel(old_tc);
deadbeefcbecd352015-09-23 11:50:27 -0700290 transport_controller_->DestroyTransportChannel_w(
291 transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTP);
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000292 }
293
294 transport_channel_ = new_tc;
295
296 if (new_tc) {
297 ConnectToTransportChannel(new_tc);
deadbeefcbecd352015-09-23 11:50:27 -0700298 for (const auto& pair : socket_options_) {
299 new_tc->SetOption(pair.first, pair.second);
300 }
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000301 }
deadbeefcbecd352015-09-23 11:50:27 -0700302
303 // Update aggregate writable/ready-to-send state between RTP and RTCP upon
304 // setting new channel
305 UpdateWritableState_w();
306 SetReadyToSend(false, new_tc && new_tc->writable());
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000307}
308
309void BaseChannel::set_rtcp_transport_channel(TransportChannel* new_tc) {
310 ASSERT(worker_thread_ == rtc::Thread::Current());
311
312 TransportChannel* old_tc = rtcp_transport_channel_;
deadbeefcbecd352015-09-23 11:50:27 -0700313 if (!old_tc && !new_tc) {
314 // Nothing to do
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000315 return;
316 }
deadbeefcbecd352015-09-23 11:50:27 -0700317 ASSERT(old_tc != new_tc);
318
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000319 if (old_tc) {
320 DisconnectFromTransportChannel(old_tc);
deadbeefcbecd352015-09-23 11:50:27 -0700321 transport_controller_->DestroyTransportChannel_w(
322 transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000323 }
324
325 rtcp_transport_channel_ = new_tc;
326
327 if (new_tc) {
Guo-wei Shieh9c38c2d2015-12-05 09:46:07 -0800328 RTC_CHECK(!(ShouldSetupDtlsSrtp() && srtp_filter_.IsActive()))
329 << "Setting RTCP for DTLS/SRTP after SrtpFilter is active "
330 << "should never happen.";
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000331 ConnectToTransportChannel(new_tc);
deadbeefcbecd352015-09-23 11:50:27 -0700332 for (const auto& pair : rtcp_socket_options_) {
333 new_tc->SetOption(pair.first, pair.second);
334 }
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000335 }
deadbeefcbecd352015-09-23 11:50:27 -0700336
337 // Update aggregate writable/ready-to-send state between RTP and RTCP upon
338 // setting new channel
339 UpdateWritableState_w();
340 SetReadyToSend(true, new_tc && new_tc->writable());
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000341}
342
343void BaseChannel::ConnectToTransportChannel(TransportChannel* tc) {
344 ASSERT(worker_thread_ == rtc::Thread::Current());
345
346 tc->SignalWritableState.connect(this, &BaseChannel::OnWritableState);
347 tc->SignalReadPacket.connect(this, &BaseChannel::OnChannelRead);
348 tc->SignalReadyToSend.connect(this, &BaseChannel::OnReadyToSend);
Guo-wei Shieh9c38c2d2015-12-05 09:46:07 -0800349 tc->SignalDtlsState.connect(this, &BaseChannel::OnDtlsState);
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000350}
351
352void BaseChannel::DisconnectFromTransportChannel(TransportChannel* tc) {
353 ASSERT(worker_thread_ == rtc::Thread::Current());
354
355 tc->SignalWritableState.disconnect(this);
356 tc->SignalReadPacket.disconnect(this);
357 tc->SignalReadyToSend.disconnect(this);
Guo-wei Shieh9c38c2d2015-12-05 09:46:07 -0800358 tc->SignalDtlsState.disconnect(this);
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +0000359}
360
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000361bool BaseChannel::Enable(bool enable) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000362 worker_thread_->Invoke<void>(Bind(
363 enable ? &BaseChannel::EnableMedia_w : &BaseChannel::DisableMedia_w,
364 this));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000365 return true;
366}
367
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000368bool BaseChannel::AddRecvStream(const StreamParams& sp) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000369 return InvokeOnWorker(Bind(&BaseChannel::AddRecvStream_w, this, sp));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000370}
371
Peter Boström0c4e06b2015-10-07 12:23:21 +0200372bool BaseChannel::RemoveRecvStream(uint32_t ssrc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000373 return InvokeOnWorker(Bind(&BaseChannel::RemoveRecvStream_w, this, ssrc));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000374}
375
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000376bool BaseChannel::AddSendStream(const StreamParams& sp) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000377 return InvokeOnWorker(
378 Bind(&MediaChannel::AddSendStream, media_channel(), sp));
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000379}
380
Peter Boström0c4e06b2015-10-07 12:23:21 +0200381bool BaseChannel::RemoveSendStream(uint32_t ssrc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000382 return InvokeOnWorker(
383 Bind(&MediaChannel::RemoveSendStream, media_channel(), ssrc));
wu@webrtc.orgcadf9042013-08-30 21:24:16 +0000384}
385
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000386bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000387 ContentAction action,
388 std::string* error_desc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000389 return InvokeOnWorker(Bind(&BaseChannel::SetLocalContent_w,
390 this, content, action, error_desc));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000391}
392
393bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000394 ContentAction action,
395 std::string* error_desc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +0000396 return InvokeOnWorker(Bind(&BaseChannel::SetRemoteContent_w,
397 this, content, action, error_desc));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000398}
399
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000400void BaseChannel::StartConnectionMonitor(int cms) {
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000401 // We pass in the BaseChannel instead of the transport_channel_
402 // because if the transport_channel_ changes, the ConnectionMonitor
403 // would be pointing to the wrong TransportChannel.
404 connection_monitor_.reset(new ConnectionMonitor(
405 this, worker_thread(), rtc::Thread::Current()));
406 connection_monitor_->SignalUpdate.connect(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000407 this, &BaseChannel::OnConnectionMonitorUpdate);
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000408 connection_monitor_->Start(cms);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000409}
410
411void BaseChannel::StopConnectionMonitor() {
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000412 if (connection_monitor_) {
413 connection_monitor_->Stop();
414 connection_monitor_.reset();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000415 }
416}
417
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +0000418bool BaseChannel::GetConnectionStats(ConnectionInfos* infos) {
419 ASSERT(worker_thread_ == rtc::Thread::Current());
420 return transport_channel_->GetStats(infos);
421}
422
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000423bool BaseChannel::IsReadyToReceive() const {
424 // Receive data if we are enabled and have local content,
425 return enabled() && IsReceiveContentDirection(local_content_direction_);
426}
427
428bool BaseChannel::IsReadyToSend() const {
429 // Send outgoing data if we are enabled, have local and remote content,
430 // and we have had some form of connectivity.
Guo-wei Shieh9c38c2d2015-12-05 09:46:07 -0800431 return enabled() && IsReceiveContentDirection(remote_content_direction_) &&
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000432 IsSendContentDirection(local_content_direction_) &&
Guo-wei Shieh9c38c2d2015-12-05 09:46:07 -0800433 was_ever_writable() &&
434 (srtp_filter_.IsActive() || !ShouldSetupDtlsSrtp());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000435}
436
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000437bool BaseChannel::SendPacket(rtc::Buffer* packet,
stefanc1aeaf02015-10-15 07:26:07 -0700438 const rtc::PacketOptions& options) {
439 return SendPacket(false, packet, options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000440}
441
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000442bool BaseChannel::SendRtcp(rtc::Buffer* packet,
stefanc1aeaf02015-10-15 07:26:07 -0700443 const rtc::PacketOptions& options) {
444 return SendPacket(true, packet, options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000445}
446
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000447int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000448 int value) {
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000449 TransportChannel* channel = NULL;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000450 switch (type) {
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000451 case ST_RTP:
452 channel = transport_channel_;
deadbeefcbecd352015-09-23 11:50:27 -0700453 socket_options_.push_back(
454 std::pair<rtc::Socket::Option, int>(opt, value));
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000455 break;
456 case ST_RTCP:
457 channel = rtcp_transport_channel_;
deadbeefcbecd352015-09-23 11:50:27 -0700458 rtcp_socket_options_.push_back(
459 std::pair<rtc::Socket::Option, int>(opt, value));
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000460 break;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000461 }
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000462 return channel ? channel->SetOption(opt, value) : -1;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000463}
464
465void BaseChannel::OnWritableState(TransportChannel* channel) {
466 ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
deadbeefcbecd352015-09-23 11:50:27 -0700467 UpdateWritableState_w();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000468}
469
470void BaseChannel::OnChannelRead(TransportChannel* channel,
wu@webrtc.orga9890802013-12-13 00:21:03 +0000471 const char* data, size_t len,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000472 const rtc::PacketTime& packet_time,
wu@webrtc.orga9890802013-12-13 00:21:03 +0000473 int flags) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000474 // OnChannelRead gets called from P2PSocket; now pass data to MediaEngine
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000475 ASSERT(worker_thread_ == rtc::Thread::Current());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000476
477 // When using RTCP multiplexing we might get RTCP packets on the RTP
478 // transport. We feed RTP traffic into the demuxer to determine if it is RTCP.
479 bool rtcp = PacketIsRtcp(channel, data, len);
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000480 rtc::Buffer packet(data, len);
wu@webrtc.orga9890802013-12-13 00:21:03 +0000481 HandlePacket(rtcp, &packet, packet_time);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000482}
483
484void BaseChannel::OnReadyToSend(TransportChannel* channel) {
deadbeefcbecd352015-09-23 11:50:27 -0700485 ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
486 SetReadyToSend(channel == rtcp_transport_channel_, true);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000487}
488
Guo-wei Shieh9c38c2d2015-12-05 09:46:07 -0800489void BaseChannel::OnDtlsState(TransportChannel* channel,
490 DtlsTransportState state) {
491 if (!ShouldSetupDtlsSrtp()) {
492 return;
493 }
494
495 // Reset the srtp filter if it's not the CONNECTED state. For the CONNECTED
496 // state, setting up DTLS-SRTP context is deferred to ChannelWritable_w to
497 // cover other scenarios like the whole channel is writable (not just this
498 // TransportChannel) or when TransportChannel is attached after DTLS is
499 // negotiated.
500 if (state != DTLS_TRANSPORT_CONNECTED) {
501 srtp_filter_.ResetParams();
502 }
503}
504
deadbeefcbecd352015-09-23 11:50:27 -0700505void BaseChannel::SetReadyToSend(bool rtcp, bool ready) {
506 if (rtcp) {
507 rtcp_ready_to_send_ = ready;
508 } else {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000509 rtp_ready_to_send_ = ready;
510 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000511
deadbeefcbecd352015-09-23 11:50:27 -0700512 if (rtp_ready_to_send_ &&
513 // In the case of rtcp mux |rtcp_transport_channel_| will be null.
514 (rtcp_ready_to_send_ || !rtcp_transport_channel_)) {
torbjornga81a42f2015-09-23 02:16:58 -0700515 // Notify the MediaChannel when both rtp and rtcp channel can send.
516 media_channel_->OnReadyToSend(true);
deadbeefcbecd352015-09-23 11:50:27 -0700517 } else {
518 // Notify the MediaChannel when either rtp or rtcp channel can't send.
519 media_channel_->OnReadyToSend(false);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000520 }
521}
522
523bool BaseChannel::PacketIsRtcp(const TransportChannel* channel,
524 const char* data, size_t len) {
525 return (channel == rtcp_transport_channel_ ||
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000526 rtcp_mux_filter_.DemuxRtcp(data, static_cast<int>(len)));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000527}
528
stefanc1aeaf02015-10-15 07:26:07 -0700529bool BaseChannel::SendPacket(bool rtcp,
530 rtc::Buffer* packet,
531 const rtc::PacketOptions& options) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000532 // SendPacket gets called from MediaEngine, typically on an encoder thread.
533 // If the thread is not our worker thread, we will post to our worker
534 // so that the real work happens on our worker. This avoids us having to
535 // synchronize access to all the pieces of the send path, including
536 // SRTP and the inner workings of the transport channels.
537 // The only downside is that we can't return a proper failure code if
538 // needed. Since UDP is unreliable anyway, this should be a non-issue.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000539 if (rtc::Thread::Current() != worker_thread_) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000540 // Avoid a copy by transferring the ownership of the packet data.
541 int message_id = (!rtcp) ? MSG_RTPPACKET : MSG_RTCPPACKET;
542 PacketMessageData* data = new PacketMessageData;
Karl Wiberg94784372015-04-20 14:03:07 +0200543 data->packet = packet->Pass();
stefanc1aeaf02015-10-15 07:26:07 -0700544 data->options = options;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000545 worker_thread_->Post(this, message_id, data);
546 return true;
547 }
548
549 // Now that we are on the correct thread, ensure we have a place to send this
550 // packet before doing anything. (We might get RTCP packets that we don't
551 // intend to send.) If we've negotiated RTCP mux, send RTCP over the RTP
552 // transport.
553 TransportChannel* channel = (!rtcp || rtcp_mux_filter_.IsActive()) ?
554 transport_channel_ : rtcp_transport_channel_;
wu@webrtc.org97077a32013-10-25 21:18:33 +0000555 if (!channel || !channel->writable()) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000556 return false;
557 }
558
559 // Protect ourselves against crazy data.
560 if (!ValidPacket(rtcp, packet)) {
561 LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " "
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000562 << PacketType(rtcp)
563 << " packet: wrong size=" << packet->size();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000564 return false;
565 }
566
stefanc1aeaf02015-10-15 07:26:07 -0700567 rtc::PacketOptions updated_options;
568 updated_options = options;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000569 // Protect if needed.
570 if (srtp_filter_.IsActive()) {
571 bool res;
Karl Wibergc56ac1e2015-05-04 14:54:55 +0200572 uint8_t* data = packet->data();
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000573 int len = static_cast<int>(packet->size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000574 if (!rtcp) {
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000575 // If ENABLE_EXTERNAL_AUTH flag is on then packet authentication is not done
576 // inside libsrtp for a RTP packet. A external HMAC module will be writing
577 // a fake HMAC value. This is ONLY done for a RTP packet.
578 // Socket layer will update rtp sendtime extension header if present in
579 // packet with current time before updating the HMAC.
580#if !defined(ENABLE_EXTERNAL_AUTH)
581 res = srtp_filter_.ProtectRtp(
582 data, len, static_cast<int>(packet->capacity()), &len);
583#else
stefanc1aeaf02015-10-15 07:26:07 -0700584 updated_options.packet_time_params.rtp_sendtime_extension_id =
henrike@webrtc.org05376342014-03-10 15:53:12 +0000585 rtp_abs_sendtime_extn_id_;
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000586 res = srtp_filter_.ProtectRtp(
587 data, len, static_cast<int>(packet->capacity()), &len,
stefanc1aeaf02015-10-15 07:26:07 -0700588 &updated_options.packet_time_params.srtp_packet_index);
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000589 // If protection succeeds, let's get auth params from srtp.
590 if (res) {
Peter Boström0c4e06b2015-10-07 12:23:21 +0200591 uint8_t* auth_key = NULL;
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000592 int key_len;
593 res = srtp_filter_.GetRtpAuthParams(
stefanc1aeaf02015-10-15 07:26:07 -0700594 &auth_key, &key_len,
595 &updated_options.packet_time_params.srtp_auth_tag_len);
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000596 if (res) {
stefanc1aeaf02015-10-15 07:26:07 -0700597 updated_options.packet_time_params.srtp_auth_key.resize(key_len);
598 updated_options.packet_time_params.srtp_auth_key.assign(
599 auth_key, auth_key + key_len);
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +0000600 }
601 }
602#endif
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000603 if (!res) {
604 int seq_num = -1;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200605 uint32_t ssrc = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000606 GetRtpSeqNum(data, len, &seq_num);
607 GetRtpSsrc(data, len, &ssrc);
608 LOG(LS_ERROR) << "Failed to protect " << content_name_
609 << " RTP packet: size=" << len
610 << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
611 return false;
612 }
613 } else {
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000614 res = srtp_filter_.ProtectRtcp(data, len,
615 static_cast<int>(packet->capacity()),
616 &len);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000617 if (!res) {
618 int type = -1;
619 GetRtcpType(data, len, &type);
620 LOG(LS_ERROR) << "Failed to protect " << content_name_
621 << " RTCP packet: size=" << len << ", type=" << type;
622 return false;
623 }
624 }
625
626 // Update the length of the packet now that we've added the auth tag.
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000627 packet->SetSize(len);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000628 } else if (secure_required_) {
629 // This is a double check for something that supposedly can't happen.
630 LOG(LS_ERROR) << "Can't send outgoing " << PacketType(rtcp)
631 << " packet when SRTP is inactive and crypto is required";
632
633 ASSERT(false);
634 return false;
635 }
636
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000637 // Bon voyage.
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000638 int ret =
stefanc1aeaf02015-10-15 07:26:07 -0700639 channel->SendPacket(packet->data<char>(), packet->size(), updated_options,
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000640 (secure() && secure_dtls()) ? PF_SRTP_BYPASS : 0);
641 if (ret != static_cast<int>(packet->size())) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000642 if (channel->GetError() == EWOULDBLOCK) {
643 LOG(LS_WARNING) << "Got EWOULDBLOCK from socket.";
deadbeefcbecd352015-09-23 11:50:27 -0700644 SetReadyToSend(rtcp, false);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000645 }
646 return false;
647 }
648 return true;
649}
650
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000651bool BaseChannel::WantsPacket(bool rtcp, rtc::Buffer* packet) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000652 // Protect ourselves against crazy data.
653 if (!ValidPacket(rtcp, packet)) {
654 LOG(LS_ERROR) << "Dropping incoming " << content_name_ << " "
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000655 << PacketType(rtcp)
656 << " packet: wrong size=" << packet->size();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000657 return false;
658 }
pbos482b12e2015-11-16 10:19:58 -0800659 if (rtcp) {
660 // Permit all (seemingly valid) RTCP packets.
661 return true;
662 }
663 // Check whether we handle this payload.
664 return bundle_filter_.DemuxPacket(packet->data<uint8_t>(), packet->size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000665}
666
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000667void BaseChannel::HandlePacket(bool rtcp, rtc::Buffer* packet,
668 const rtc::PacketTime& packet_time) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000669 if (!WantsPacket(rtcp, packet)) {
670 return;
671 }
672
honghaiz@google.coma67ca1a2015-01-28 19:48:33 +0000673 // We are only interested in the first rtp packet because that
674 // indicates the media has started flowing.
675 if (!has_received_packet_ && !rtcp) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000676 has_received_packet_ = true;
677 signaling_thread()->Post(this, MSG_FIRSTPACKETRECEIVED);
678 }
679
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000680 // Unprotect the packet, if needed.
681 if (srtp_filter_.IsActive()) {
Karl Wiberg94784372015-04-20 14:03:07 +0200682 char* data = packet->data<char>();
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000683 int len = static_cast<int>(packet->size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000684 bool res;
685 if (!rtcp) {
686 res = srtp_filter_.UnprotectRtp(data, len, &len);
687 if (!res) {
688 int seq_num = -1;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200689 uint32_t ssrc = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000690 GetRtpSeqNum(data, len, &seq_num);
691 GetRtpSsrc(data, len, &ssrc);
692 LOG(LS_ERROR) << "Failed to unprotect " << content_name_
693 << " RTP packet: size=" << len
694 << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
695 return;
696 }
697 } else {
698 res = srtp_filter_.UnprotectRtcp(data, len, &len);
699 if (!res) {
700 int type = -1;
701 GetRtcpType(data, len, &type);
702 LOG(LS_ERROR) << "Failed to unprotect " << content_name_
703 << " RTCP packet: size=" << len << ", type=" << type;
704 return;
705 }
706 }
707
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +0000708 packet->SetSize(len);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000709 } else if (secure_required_) {
710 // Our session description indicates that SRTP is required, but we got a
711 // packet before our SRTP filter is active. This means either that
712 // a) we got SRTP packets before we received the SDES keys, in which case
713 // we can't decrypt it anyway, or
714 // b) we got SRTP packets before DTLS completed on both the RTP and RTCP
715 // channels, so we haven't yet extracted keys, even if DTLS did complete
716 // on the channel that the packets are being sent on. It's really good
717 // practice to wait for both RTP and RTCP to be good to go before sending
718 // media, to prevent weird failure modes, so it's fine for us to just eat
719 // packets here. This is all sidestepped if RTCP mux is used anyway.
720 LOG(LS_WARNING) << "Can't process incoming " << PacketType(rtcp)
721 << " packet when SRTP is inactive and crypto is required";
722 return;
723 }
724
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000725 // Push it down to the media channel.
726 if (!rtcp) {
wu@webrtc.orga9890802013-12-13 00:21:03 +0000727 media_channel_->OnPacketReceived(packet, packet_time);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000728 } else {
wu@webrtc.orga9890802013-12-13 00:21:03 +0000729 media_channel_->OnRtcpReceived(packet, packet_time);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000730 }
731}
732
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000733bool BaseChannel::PushdownLocalDescription(
734 const SessionDescription* local_desc, ContentAction action,
735 std::string* error_desc) {
736 const ContentInfo* content_info = GetFirstContent(local_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000737 const MediaContentDescription* content_desc =
738 GetContentDescription(content_info);
739 if (content_desc && content_info && !content_info->rejected &&
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000740 !SetLocalContent(content_desc, action, error_desc)) {
741 LOG(LS_ERROR) << "Failure in SetLocalContent with action " << action;
742 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000743 }
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000744 return true;
745}
746
747bool BaseChannel::PushdownRemoteDescription(
748 const SessionDescription* remote_desc, ContentAction action,
749 std::string* error_desc) {
750 const ContentInfo* content_info = GetFirstContent(remote_desc);
751 const MediaContentDescription* content_desc =
752 GetContentDescription(content_info);
753 if (content_desc && content_info && !content_info->rejected &&
754 !SetRemoteContent(content_desc, action, error_desc)) {
755 LOG(LS_ERROR) << "Failure in SetRemoteContent with action " << action;
756 return false;
757 }
758 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000759}
760
761void BaseChannel::EnableMedia_w() {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000762 ASSERT(worker_thread_ == rtc::Thread::Current());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000763 if (enabled_)
764 return;
765
766 LOG(LS_INFO) << "Channel enabled";
767 enabled_ = true;
768 ChangeState();
769}
770
771void BaseChannel::DisableMedia_w() {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000772 ASSERT(worker_thread_ == rtc::Thread::Current());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000773 if (!enabled_)
774 return;
775
776 LOG(LS_INFO) << "Channel disabled";
777 enabled_ = false;
778 ChangeState();
779}
780
deadbeefcbecd352015-09-23 11:50:27 -0700781void BaseChannel::UpdateWritableState_w() {
782 if (transport_channel_ && transport_channel_->writable() &&
783 (!rtcp_transport_channel_ || rtcp_transport_channel_->writable())) {
784 ChannelWritable_w();
785 } else {
786 ChannelNotWritable_w();
787 }
788}
789
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000790void BaseChannel::ChannelWritable_w() {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000791 ASSERT(worker_thread_ == rtc::Thread::Current());
Guo-wei Shieh9c38c2d2015-12-05 09:46:07 -0800792 if (writable_) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000793 return;
Guo-wei Shieh9c38c2d2015-12-05 09:46:07 -0800794 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000795
deadbeefcbecd352015-09-23 11:50:27 -0700796 LOG(LS_INFO) << "Channel writable (" << content_name_ << ")"
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000797 << (was_ever_writable_ ? "" : " for the first time");
798
799 std::vector<ConnectionInfo> infos;
800 transport_channel_->GetStats(&infos);
801 for (std::vector<ConnectionInfo>::const_iterator it = infos.begin();
802 it != infos.end(); ++it) {
803 if (it->best_connection) {
804 LOG(LS_INFO) << "Using " << it->local_candidate.ToSensitiveString()
805 << "->" << it->remote_candidate.ToSensitiveString();
806 break;
807 }
808 }
809
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000810 was_ever_writable_ = true;
Guo-wei Shieh9c38c2d2015-12-05 09:46:07 -0800811 MaybeSetupDtlsSrtp_w();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000812 writable_ = true;
813 ChangeState();
814}
815
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +0000816void BaseChannel::SignalDtlsSetupFailure_w(bool rtcp) {
817 ASSERT(worker_thread() == rtc::Thread::Current());
818 signaling_thread()->Invoke<void>(Bind(
819 &BaseChannel::SignalDtlsSetupFailure_s, this, rtcp));
820}
821
822void BaseChannel::SignalDtlsSetupFailure_s(bool rtcp) {
823 ASSERT(signaling_thread() == rtc::Thread::Current());
824 SignalDtlsSetupFailure(this, rtcp);
825}
826
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800827bool BaseChannel::SetDtlsSrtpCryptoSuites(TransportChannel* tc, bool rtcp) {
828 std::vector<int> crypto_suites;
829 // We always use the default SRTP crypto suites for RTCP, but we may use
830 // different crypto suites for RTP depending on the media type.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000831 if (!rtcp) {
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800832 GetSrtpCryptoSuites(&crypto_suites);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000833 } else {
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800834 GetDefaultSrtpCryptoSuites(&crypto_suites);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000835 }
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800836 return tc->SetSrtpCryptoSuites(crypto_suites);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000837}
838
839bool BaseChannel::ShouldSetupDtlsSrtp() const {
Guo-wei Shieh9c38c2d2015-12-05 09:46:07 -0800840 // Since DTLS is applied to all channels, checking RTP should be enough.
841 return transport_channel_ && transport_channel_->IsDtlsActive();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000842}
843
844// This function returns true if either DTLS-SRTP is not in use
845// *or* DTLS-SRTP is successfully set up.
846bool BaseChannel::SetupDtlsSrtp(bool rtcp_channel) {
847 bool ret = false;
848
deadbeefcbecd352015-09-23 11:50:27 -0700849 TransportChannel* channel =
850 rtcp_channel ? rtcp_transport_channel_ : transport_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000851
Guo-wei Shieh9c38c2d2015-12-05 09:46:07 -0800852 RTC_DCHECK(channel->IsDtlsActive());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000853
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800854 int selected_crypto_suite;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000855
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800856 if (!channel->GetSrtpCryptoSuite(&selected_crypto_suite)) {
857 LOG(LS_ERROR) << "No DTLS-SRTP selected crypto suite";
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000858 return false;
859 }
860
861 LOG(LS_INFO) << "Installing keys from DTLS-SRTP on "
862 << content_name() << " "
863 << PacketType(rtcp_channel);
864
865 // OK, we're now doing DTLS (RFC 5764)
866 std::vector<unsigned char> dtls_buffer(SRTP_MASTER_KEY_KEY_LEN * 2 +
867 SRTP_MASTER_KEY_SALT_LEN * 2);
868
869 // RFC 5705 exporter using the RFC 5764 parameters
870 if (!channel->ExportKeyingMaterial(
871 kDtlsSrtpExporterLabel,
872 NULL, 0, false,
873 &dtls_buffer[0], dtls_buffer.size())) {
874 LOG(LS_WARNING) << "DTLS-SRTP key export failed";
875 ASSERT(false); // This should never happen
876 return false;
877 }
878
879 // Sync up the keys with the DTLS-SRTP interface
880 std::vector<unsigned char> client_write_key(SRTP_MASTER_KEY_KEY_LEN +
881 SRTP_MASTER_KEY_SALT_LEN);
882 std::vector<unsigned char> server_write_key(SRTP_MASTER_KEY_KEY_LEN +
883 SRTP_MASTER_KEY_SALT_LEN);
884 size_t offset = 0;
885 memcpy(&client_write_key[0], &dtls_buffer[offset],
886 SRTP_MASTER_KEY_KEY_LEN);
887 offset += SRTP_MASTER_KEY_KEY_LEN;
888 memcpy(&server_write_key[0], &dtls_buffer[offset],
889 SRTP_MASTER_KEY_KEY_LEN);
890 offset += SRTP_MASTER_KEY_KEY_LEN;
891 memcpy(&client_write_key[SRTP_MASTER_KEY_KEY_LEN],
892 &dtls_buffer[offset], SRTP_MASTER_KEY_SALT_LEN);
893 offset += SRTP_MASTER_KEY_SALT_LEN;
894 memcpy(&server_write_key[SRTP_MASTER_KEY_KEY_LEN],
895 &dtls_buffer[offset], SRTP_MASTER_KEY_SALT_LEN);
896
897 std::vector<unsigned char> *send_key, *recv_key;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000898 rtc::SSLRole role;
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000899 if (!channel->GetSslRole(&role)) {
900 LOG(LS_WARNING) << "GetSslRole failed";
901 return false;
902 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000903
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000904 if (role == rtc::SSL_SERVER) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000905 send_key = &server_write_key;
906 recv_key = &client_write_key;
907 } else {
908 send_key = &client_write_key;
909 recv_key = &server_write_key;
910 }
911
912 if (rtcp_channel) {
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800913 ret = srtp_filter_.SetRtcpParams(selected_crypto_suite, &(*send_key)[0],
914 static_cast<int>(send_key->size()),
915 selected_crypto_suite, &(*recv_key)[0],
916 static_cast<int>(recv_key->size()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000917 } else {
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -0800918 ret = srtp_filter_.SetRtpParams(selected_crypto_suite, &(*send_key)[0],
919 static_cast<int>(send_key->size()),
920 selected_crypto_suite, &(*recv_key)[0],
921 static_cast<int>(recv_key->size()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000922 }
923
924 if (!ret)
925 LOG(LS_WARNING) << "DTLS-SRTP key installation failed";
926 else
927 dtls_keyed_ = true;
928
929 return ret;
930}
931
Guo-wei Shieh9c38c2d2015-12-05 09:46:07 -0800932void BaseChannel::MaybeSetupDtlsSrtp_w() {
933 if (srtp_filter_.IsActive()) {
934 return;
935 }
936
937 if (!ShouldSetupDtlsSrtp()) {
938 return;
939 }
940
941 if (!SetupDtlsSrtp(false)) {
942 SignalDtlsSetupFailure_w(false);
943 return;
944 }
945
946 if (rtcp_transport_channel_) {
947 if (!SetupDtlsSrtp(true)) {
948 SignalDtlsSetupFailure_w(true);
949 return;
950 }
951 }
952}
953
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000954void BaseChannel::ChannelNotWritable_w() {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000955 ASSERT(worker_thread_ == rtc::Thread::Current());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000956 if (!writable_)
957 return;
958
deadbeefcbecd352015-09-23 11:50:27 -0700959 LOG(LS_INFO) << "Channel not writable (" << content_name_ << ")";
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000960 writable_ = false;
961 ChangeState();
962}
963
Peter Thatcherc2ee2c82015-08-07 16:05:34 -0700964bool BaseChannel::SetRtpTransportParameters_w(
965 const MediaContentDescription* content,
966 ContentAction action,
967 ContentSource src,
968 std::string* error_desc) {
969 if (action == CA_UPDATE) {
970 // These parameters never get changed by a CA_UDPATE.
971 return true;
972 }
973
974 // Cache secure_required_ for belt and suspenders check on SendPacket
975 if (src == CS_LOCAL) {
976 set_secure_required(content->crypto_required() != CT_NONE);
977 }
978
979 if (!SetSrtp_w(content->cryptos(), action, src, error_desc)) {
980 return false;
981 }
982
983 if (!SetRtcpMux_w(content->rtcp_mux(), action, src, error_desc)) {
984 return false;
985 }
986
987 return true;
988}
989
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000990// |dtls| will be set to true if DTLS is active for transport channel and
991// crypto is empty.
992bool BaseChannel::CheckSrtpConfig(const std::vector<CryptoParams>& cryptos,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000993 bool* dtls,
994 std::string* error_desc) {
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000995 *dtls = transport_channel_->IsDtlsActive();
996 if (*dtls && !cryptos.empty()) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000997 SafeSetError("Cryptos must be empty when DTLS is active.",
998 error_desc);
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +0000999 return false;
1000 }
1001 return true;
1002}
1003
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001004bool BaseChannel::SetSrtp_w(const std::vector<CryptoParams>& cryptos,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001005 ContentAction action,
1006 ContentSource src,
1007 std::string* error_desc) {
1008 if (action == CA_UPDATE) {
1009 // no crypto params.
1010 return true;
1011 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001012 bool ret = false;
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00001013 bool dtls = false;
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001014 ret = CheckSrtpConfig(cryptos, &dtls, error_desc);
1015 if (!ret) {
1016 return false;
1017 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001018 switch (action) {
1019 case CA_OFFER:
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00001020 // If DTLS is already active on the channel, we could be renegotiating
1021 // here. We don't update the srtp filter.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001022 if (!dtls) {
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +00001023 ret = srtp_filter_.SetOffer(cryptos, src);
1024 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001025 break;
1026 case CA_PRANSWER:
1027 // If we're doing DTLS-SRTP, we don't want to update the filter
1028 // with an answer, because we already have SRTP parameters.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001029 if (!dtls) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001030 ret = srtp_filter_.SetProvisionalAnswer(cryptos, src);
1031 }
1032 break;
1033 case CA_ANSWER:
1034 // If we're doing DTLS-SRTP, we don't want to update the filter
1035 // with an answer, because we already have SRTP parameters.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001036 if (!dtls) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001037 ret = srtp_filter_.SetAnswer(cryptos, src);
1038 }
1039 break;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001040 default:
1041 break;
1042 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001043 if (!ret) {
1044 SafeSetError("Failed to setup SRTP filter.", error_desc);
1045 return false;
1046 }
1047 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001048}
1049
Peter Thatcheraf55ccc2015-05-21 07:48:41 -07001050void BaseChannel::ActivateRtcpMux() {
1051 worker_thread_->Invoke<void>(Bind(
1052 &BaseChannel::ActivateRtcpMux_w, this));
1053}
1054
1055void BaseChannel::ActivateRtcpMux_w() {
1056 if (!rtcp_mux_filter_.IsActive()) {
1057 rtcp_mux_filter_.SetActive();
deadbeefcbecd352015-09-23 11:50:27 -07001058 set_rtcp_transport_channel(nullptr);
1059 rtcp_transport_enabled_ = false;
Peter Thatcheraf55ccc2015-05-21 07:48:41 -07001060 }
1061}
1062
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001063bool BaseChannel::SetRtcpMux_w(bool enable, ContentAction action,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001064 ContentSource src,
1065 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001066 bool ret = false;
1067 switch (action) {
1068 case CA_OFFER:
1069 ret = rtcp_mux_filter_.SetOffer(enable, src);
1070 break;
1071 case CA_PRANSWER:
1072 ret = rtcp_mux_filter_.SetProvisionalAnswer(enable, src);
1073 break;
1074 case CA_ANSWER:
1075 ret = rtcp_mux_filter_.SetAnswer(enable, src);
1076 if (ret && rtcp_mux_filter_.IsActive()) {
1077 // We activated RTCP mux, close down the RTCP transport.
deadbeefcbecd352015-09-23 11:50:27 -07001078 LOG(LS_INFO) << "Enabling rtcp-mux for " << content_name()
1079 << " by destroying RTCP transport channel for "
1080 << transport_name();
1081 set_rtcp_transport_channel(nullptr);
1082 rtcp_transport_enabled_ = false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001083 }
1084 break;
1085 case CA_UPDATE:
1086 // No RTCP mux info.
1087 ret = true;
Henrik Kjellander7c027b62015-04-22 13:21:30 +02001088 break;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001089 default:
1090 break;
1091 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001092 if (!ret) {
1093 SafeSetError("Failed to setup RTCP mux filter.", error_desc);
1094 return false;
1095 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001096 // |rtcp_mux_filter_| can be active if |action| is CA_PRANSWER or
1097 // CA_ANSWER, but we only want to tear down the RTCP transport channel if we
1098 // received a final answer.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001099 if (rtcp_mux_filter_.IsActive()) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001100 // If the RTP transport is already writable, then so are we.
1101 if (transport_channel_->writable()) {
1102 ChannelWritable_w();
1103 }
1104 }
1105
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001106 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001107}
1108
1109bool BaseChannel::AddRecvStream_w(const StreamParams& sp) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001110 ASSERT(worker_thread() == rtc::Thread::Current());
pbos482b12e2015-11-16 10:19:58 -08001111 return media_channel()->AddRecvStream(sp);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001112}
1113
Peter Boström0c4e06b2015-10-07 12:23:21 +02001114bool BaseChannel::RemoveRecvStream_w(uint32_t ssrc) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001115 ASSERT(worker_thread() == rtc::Thread::Current());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001116 return media_channel()->RemoveRecvStream(ssrc);
1117}
1118
1119bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001120 ContentAction action,
1121 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001122 if (!VERIFY(action == CA_OFFER || action == CA_ANSWER ||
1123 action == CA_PRANSWER || action == CA_UPDATE))
1124 return false;
1125
1126 // If this is an update, streams only contain streams that have changed.
1127 if (action == CA_UPDATE) {
1128 for (StreamParamsVec::const_iterator it = streams.begin();
1129 it != streams.end(); ++it) {
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +00001130 const StreamParams* existing_stream =
1131 GetStreamByIds(local_streams_, it->groupid, it->id);
1132 if (!existing_stream && it->has_ssrcs()) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001133 if (media_channel()->AddSendStream(*it)) {
1134 local_streams_.push_back(*it);
1135 LOG(LS_INFO) << "Add send stream ssrc: " << it->first_ssrc();
1136 } else {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001137 std::ostringstream desc;
1138 desc << "Failed to add send stream ssrc: " << it->first_ssrc();
1139 SafeSetError(desc.str(), error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001140 return false;
1141 }
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +00001142 } else if (existing_stream && !it->has_ssrcs()) {
1143 if (!media_channel()->RemoveSendStream(existing_stream->first_ssrc())) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001144 std::ostringstream desc;
1145 desc << "Failed to remove send stream with ssrc "
1146 << it->first_ssrc() << ".";
1147 SafeSetError(desc.str(), error_desc);
1148 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001149 }
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +00001150 RemoveStreamBySsrc(&local_streams_, existing_stream->first_ssrc());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001151 } else {
1152 LOG(LS_WARNING) << "Ignore unsupported stream update";
1153 }
1154 }
1155 return true;
1156 }
1157 // Else streams are all the streams we want to send.
1158
1159 // Check for streams that have been removed.
1160 bool ret = true;
1161 for (StreamParamsVec::const_iterator it = local_streams_.begin();
1162 it != local_streams_.end(); ++it) {
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +00001163 if (!GetStreamBySsrc(streams, it->first_ssrc())) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001164 if (!media_channel()->RemoveSendStream(it->first_ssrc())) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001165 std::ostringstream desc;
1166 desc << "Failed to remove send stream with ssrc "
1167 << it->first_ssrc() << ".";
1168 SafeSetError(desc.str(), error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001169 ret = false;
1170 }
1171 }
1172 }
1173 // Check for new streams.
1174 for (StreamParamsVec::const_iterator it = streams.begin();
1175 it != streams.end(); ++it) {
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +00001176 if (!GetStreamBySsrc(local_streams_, it->first_ssrc())) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001177 if (media_channel()->AddSendStream(*it)) {
stefanc1aeaf02015-10-15 07:26:07 -07001178 LOG(LS_INFO) << "Add send stream ssrc: " << it->ssrcs[0];
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001179 } else {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001180 std::ostringstream desc;
1181 desc << "Failed to add send stream ssrc: " << it->first_ssrc();
1182 SafeSetError(desc.str(), error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001183 ret = false;
1184 }
1185 }
1186 }
1187 local_streams_ = streams;
1188 return ret;
1189}
1190
1191bool BaseChannel::UpdateRemoteStreams_w(
1192 const std::vector<StreamParams>& streams,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001193 ContentAction action,
1194 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001195 if (!VERIFY(action == CA_OFFER || action == CA_ANSWER ||
1196 action == CA_PRANSWER || action == CA_UPDATE))
1197 return false;
1198
1199 // If this is an update, streams only contain streams that have changed.
1200 if (action == CA_UPDATE) {
1201 for (StreamParamsVec::const_iterator it = streams.begin();
1202 it != streams.end(); ++it) {
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +00001203 const StreamParams* existing_stream =
1204 GetStreamByIds(remote_streams_, it->groupid, it->id);
1205 if (!existing_stream && it->has_ssrcs()) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001206 if (AddRecvStream_w(*it)) {
1207 remote_streams_.push_back(*it);
1208 LOG(LS_INFO) << "Add remote stream ssrc: " << it->first_ssrc();
1209 } else {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001210 std::ostringstream desc;
1211 desc << "Failed to add remote stream ssrc: " << it->first_ssrc();
1212 SafeSetError(desc.str(), error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001213 return false;
1214 }
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +00001215 } else if (existing_stream && !it->has_ssrcs()) {
1216 if (!RemoveRecvStream_w(existing_stream->first_ssrc())) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001217 std::ostringstream desc;
1218 desc << "Failed to remove remote stream with ssrc "
1219 << it->first_ssrc() << ".";
1220 SafeSetError(desc.str(), error_desc);
1221 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001222 }
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +00001223 RemoveStreamBySsrc(&remote_streams_, existing_stream->first_ssrc());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001224 } else {
1225 LOG(LS_WARNING) << "Ignore unsupported stream update."
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +00001226 << " Stream exists? " << (existing_stream != nullptr)
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001227 << " new stream = " << it->ToString();
1228 }
1229 }
1230 return true;
1231 }
1232 // Else streams are all the streams we want to receive.
1233
1234 // Check for streams that have been removed.
1235 bool ret = true;
1236 for (StreamParamsVec::const_iterator it = remote_streams_.begin();
1237 it != remote_streams_.end(); ++it) {
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +00001238 if (!GetStreamBySsrc(streams, it->first_ssrc())) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001239 if (!RemoveRecvStream_w(it->first_ssrc())) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001240 std::ostringstream desc;
1241 desc << "Failed to remove remote stream with ssrc "
1242 << it->first_ssrc() << ".";
1243 SafeSetError(desc.str(), error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001244 ret = false;
1245 }
1246 }
1247 }
1248 // Check for new streams.
1249 for (StreamParamsVec::const_iterator it = streams.begin();
1250 it != streams.end(); ++it) {
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +00001251 if (!GetStreamBySsrc(remote_streams_, it->first_ssrc())) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001252 if (AddRecvStream_w(*it)) {
1253 LOG(LS_INFO) << "Add remote ssrc: " << it->ssrcs[0];
1254 } else {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001255 std::ostringstream desc;
1256 desc << "Failed to add remote stream ssrc: " << it->first_ssrc();
1257 SafeSetError(desc.str(), error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001258 ret = false;
1259 }
1260 }
1261 }
1262 remote_streams_ = streams;
1263 return ret;
1264}
1265
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +00001266void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension(
1267 const std::vector<RtpHeaderExtension>& extensions) {
1268 const RtpHeaderExtension* send_time_extension =
henrike@webrtc.org79047f92014-03-06 23:46:59 +00001269 FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
henrike@webrtc.orgd43aa9d2014-02-21 23:43:24 +00001270 rtp_abs_sendtime_extn_id_ =
1271 send_time_extension ? send_time_extension->id : -1;
1272}
1273
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001274void BaseChannel::OnMessage(rtc::Message *pmsg) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001275 switch (pmsg->message_id) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001276 case MSG_RTPPACKET:
1277 case MSG_RTCPPACKET: {
1278 PacketMessageData* data = static_cast<PacketMessageData*>(pmsg->pdata);
stefanc1aeaf02015-10-15 07:26:07 -07001279 SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet,
1280 data->options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001281 delete data; // because it is Posted
1282 break;
1283 }
1284 case MSG_FIRSTPACKETRECEIVED: {
1285 SignalFirstPacketReceived(this);
1286 break;
1287 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001288 }
1289}
1290
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001291void BaseChannel::FlushRtcpMessages() {
1292 // Flush all remaining RTCP messages. This should only be called in
1293 // destructor.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001294 ASSERT(rtc::Thread::Current() == worker_thread_);
1295 rtc::MessageList rtcp_messages;
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001296 worker_thread_->Clear(this, MSG_RTCPPACKET, &rtcp_messages);
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001297 for (rtc::MessageList::iterator it = rtcp_messages.begin();
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001298 it != rtcp_messages.end(); ++it) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001299 worker_thread_->Send(this, MSG_RTCPPACKET, it->pdata);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001300 }
1301}
1302
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001303VoiceChannel::VoiceChannel(rtc::Thread* thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001304 MediaEngineInterface* media_engine,
1305 VoiceMediaChannel* media_channel,
deadbeefcbecd352015-09-23 11:50:27 -07001306 TransportController* transport_controller,
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001307 const std::string& content_name,
1308 bool rtcp)
deadbeefcbecd352015-09-23 11:50:27 -07001309 : BaseChannel(thread,
1310 media_channel,
1311 transport_controller,
1312 content_name,
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001313 rtcp),
Fredrik Solenberg0c022642015-08-05 12:25:22 +02001314 media_engine_(media_engine),
deadbeefcbecd352015-09-23 11:50:27 -07001315 received_media_(false) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001316
1317VoiceChannel::~VoiceChannel() {
1318 StopAudioMonitor();
1319 StopMediaMonitor();
1320 // this can't be done in the base class, since it calls a virtual
1321 DisableMedia_w();
wu@webrtc.org78187522013-10-07 23:32:02 +00001322 Deinit();
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001323}
1324
1325bool VoiceChannel::Init() {
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +00001326 if (!BaseChannel::Init()) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001327 return false;
1328 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001329 return true;
1330}
1331
Peter Boström0c4e06b2015-10-07 12:23:21 +02001332bool VoiceChannel::SetAudioSend(uint32_t ssrc,
solenbergdfc8f4f2015-10-01 02:31:10 -07001333 bool enable,
solenberg1dd98f32015-09-10 01:57:14 -07001334 const AudioOptions* options,
1335 AudioRenderer* renderer) {
deadbeefcbecd352015-09-23 11:50:27 -07001336 return InvokeOnWorker(Bind(&VoiceMediaChannel::SetAudioSend, media_channel(),
solenbergdfc8f4f2015-10-01 02:31:10 -07001337 ssrc, enable, options, renderer));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001338}
1339
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001340// TODO(juberti): Handle early media the right way. We should get an explicit
1341// ringing message telling us to start playing local ringback, which we cancel
1342// if any early media actually arrives. For now, we do the opposite, which is
1343// to wait 1 second for early media, and start playing local ringback if none
1344// arrives.
1345void VoiceChannel::SetEarlyMedia(bool enable) {
1346 if (enable) {
1347 // Start the early media timeout
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001348 worker_thread()->PostDelayed(kEarlyMediaTimeout, this,
1349 MSG_EARLYMEDIATIMEOUT);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001350 } else {
1351 // Stop the timeout if currently going.
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001352 worker_thread()->Clear(this, MSG_EARLYMEDIATIMEOUT);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001353 }
1354}
1355
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001356bool VoiceChannel::CanInsertDtmf() {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001357 return InvokeOnWorker(Bind(&VoiceMediaChannel::CanInsertDtmf,
1358 media_channel()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001359}
1360
Peter Boström0c4e06b2015-10-07 12:23:21 +02001361bool VoiceChannel::InsertDtmf(uint32_t ssrc,
1362 int event_code,
solenberg1d63dd02015-12-02 12:35:09 -08001363 int duration) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001364 return InvokeOnWorker(Bind(&VoiceChannel::InsertDtmf_w, this,
solenberg1d63dd02015-12-02 12:35:09 -08001365 ssrc, event_code, duration));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001366}
1367
solenberg4bac9c52015-10-09 02:32:53 -07001368bool VoiceChannel::SetOutputVolume(uint32_t ssrc, double volume) {
1369 return InvokeOnWorker(Bind(&VoiceMediaChannel::SetOutputVolume,
1370 media_channel(), ssrc, volume));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001371}
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001372
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001373bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001374 return InvokeOnWorker(Bind(&VoiceMediaChannel::GetStats,
1375 media_channel(), stats));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001376}
1377
1378void VoiceChannel::StartMediaMonitor(int cms) {
1379 media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(),
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001380 rtc::Thread::Current()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001381 media_monitor_->SignalUpdate.connect(
1382 this, &VoiceChannel::OnMediaMonitorUpdate);
1383 media_monitor_->Start(cms);
1384}
1385
1386void VoiceChannel::StopMediaMonitor() {
1387 if (media_monitor_) {
1388 media_monitor_->Stop();
1389 media_monitor_->SignalUpdate.disconnect(this);
1390 media_monitor_.reset();
1391 }
1392}
1393
1394void VoiceChannel::StartAudioMonitor(int cms) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001395 audio_monitor_.reset(new AudioMonitor(this, rtc::Thread::Current()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001396 audio_monitor_
1397 ->SignalUpdate.connect(this, &VoiceChannel::OnAudioMonitorUpdate);
1398 audio_monitor_->Start(cms);
1399}
1400
1401void VoiceChannel::StopAudioMonitor() {
1402 if (audio_monitor_) {
1403 audio_monitor_->Stop();
1404 audio_monitor_.reset();
1405 }
1406}
1407
1408bool VoiceChannel::IsAudioMonitorRunning() const {
1409 return (audio_monitor_.get() != NULL);
1410}
1411
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001412int VoiceChannel::GetInputLevel_w() {
Fredrik Solenberg0c022642015-08-05 12:25:22 +02001413 return media_engine_->GetInputLevel();
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001414}
1415
1416int VoiceChannel::GetOutputLevel_w() {
1417 return media_channel()->GetOutputLevel();
1418}
1419
1420void VoiceChannel::GetActiveStreams_w(AudioInfo::StreamList* actives) {
1421 media_channel()->GetActiveStreams(actives);
1422}
1423
1424void VoiceChannel::OnChannelRead(TransportChannel* channel,
wu@webrtc.orga9890802013-12-13 00:21:03 +00001425 const char* data, size_t len,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001426 const rtc::PacketTime& packet_time,
wu@webrtc.orga9890802013-12-13 00:21:03 +00001427 int flags) {
1428 BaseChannel::OnChannelRead(channel, data, len, packet_time, flags);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001429
1430 // Set a flag when we've received an RTP packet. If we're waiting for early
1431 // media, this will disable the timeout.
1432 if (!received_media_ && !PacketIsRtcp(channel, data, len)) {
1433 received_media_ = true;
1434 }
1435}
1436
1437void VoiceChannel::ChangeState() {
1438 // Render incoming data if we're the active call, and we have the local
1439 // content. We receive data on the default channel and multiplexed streams.
1440 bool recv = IsReadyToReceive();
solenberg5b14b422015-10-01 04:10:31 -07001441 media_channel()->SetPlayout(recv);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001442
1443 // Send outgoing data if we're the active call, we have the remote content,
1444 // and we have had some form of connectivity.
1445 bool send = IsReadyToSend();
1446 SendFlags send_flag = send ? SEND_MICROPHONE : SEND_NOTHING;
1447 if (!media_channel()->SetSend(send_flag)) {
1448 LOG(LS_ERROR) << "Failed to SetSend " << send_flag << " on voice channel";
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001449 }
1450
1451 LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send;
1452}
1453
1454const ContentInfo* VoiceChannel::GetFirstContent(
1455 const SessionDescription* sdesc) {
1456 return GetFirstAudioContent(sdesc);
1457}
1458
1459bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001460 ContentAction action,
1461 std::string* error_desc) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001462 ASSERT(worker_thread() == rtc::Thread::Current());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001463 LOG(LS_INFO) << "Setting local voice description";
1464
1465 const AudioContentDescription* audio =
1466 static_cast<const AudioContentDescription*>(content);
1467 ASSERT(audio != NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001468 if (!audio) {
1469 SafeSetError("Can't find audio content in local description.", error_desc);
1470 return false;
1471 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001472
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001473 if (!SetRtpTransportParameters_w(content, action, CS_LOCAL, error_desc)) {
1474 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001475 }
1476
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001477 AudioRecvParameters recv_params = last_recv_params_;
1478 RtpParametersFromMediaDescription(audio, &recv_params);
1479 if (!media_channel()->SetRecvParameters(recv_params)) {
Peter Thatcherbfab5cb2015-08-20 17:40:24 -07001480 SafeSetError("Failed to set local audio description recv parameters.",
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001481 error_desc);
1482 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001483 }
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001484 for (const AudioCodec& codec : audio->codecs()) {
1485 bundle_filter()->AddPayloadType(codec.id);
1486 }
1487 last_recv_params_ = recv_params;
1488
1489 // TODO(pthatcher): Move local streams into AudioSendParameters, and
1490 // only give it to the media channel once we have a remote
1491 // description too (without a remote description, we won't be able
1492 // to send them anyway).
1493 if (!UpdateLocalStreams_w(audio->streams(), action, error_desc)) {
1494 SafeSetError("Failed to set local audio description streams.", error_desc);
1495 return false;
1496 }
1497
1498 set_local_content_direction(content->direction());
1499 ChangeState();
1500 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001501}
1502
1503bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001504 ContentAction action,
1505 std::string* error_desc) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001506 ASSERT(worker_thread() == rtc::Thread::Current());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001507 LOG(LS_INFO) << "Setting remote voice description";
1508
1509 const AudioContentDescription* audio =
1510 static_cast<const AudioContentDescription*>(content);
1511 ASSERT(audio != NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001512 if (!audio) {
1513 SafeSetError("Can't find audio content in remote description.", error_desc);
1514 return false;
1515 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001516
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001517 if (!SetRtpTransportParameters_w(content, action, CS_REMOTE, error_desc)) {
1518 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001519 }
1520
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001521 AudioSendParameters send_params = last_send_params_;
1522 RtpSendParametersFromMediaDescription(audio, &send_params);
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001523 if (audio->agc_minus_10db()) {
Karl Wibergbe579832015-11-10 22:34:18 +01001524 send_params.options.adjust_agc_delta = rtc::Optional<int>(kAgcMinus10db);
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001525 }
1526 if (!media_channel()->SetSendParameters(send_params)) {
1527 SafeSetError("Failed to set remote audio description send parameters.",
1528 error_desc);
1529 return false;
1530 }
1531 last_send_params_ = send_params;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001532
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001533 // TODO(pthatcher): Move remote streams into AudioRecvParameters,
1534 // and only give it to the media channel once we have a local
1535 // description too (without a local description, we won't be able to
1536 // recv them anyway).
1537 if (!UpdateRemoteStreams_w(audio->streams(), action, error_desc)) {
1538 SafeSetError("Failed to set remote audio description streams.", error_desc);
1539 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001540 }
1541
Peter Thatcherbfab5cb2015-08-20 17:40:24 -07001542 if (audio->rtp_header_extensions_set()) {
1543 MaybeCacheRtpAbsSendTimeHeaderExtension(audio->rtp_header_extensions());
1544 }
1545
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001546 set_remote_content_direction(content->direction());
1547 ChangeState();
1548 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001549}
1550
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001551void VoiceChannel::HandleEarlyMediaTimeout() {
1552 // This occurs on the main thread, not the worker thread.
1553 if (!received_media_) {
1554 LOG(LS_INFO) << "No early media received before timeout";
1555 SignalEarlyMediaTimeout(this);
1556 }
1557}
1558
Peter Boström0c4e06b2015-10-07 12:23:21 +02001559bool VoiceChannel::InsertDtmf_w(uint32_t ssrc,
1560 int event,
solenberg1d63dd02015-12-02 12:35:09 -08001561 int duration) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001562 if (!enabled()) {
1563 return false;
1564 }
solenberg1d63dd02015-12-02 12:35:09 -08001565 return media_channel()->InsertDtmf(ssrc, event, duration);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001566}
1567
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001568void VoiceChannel::OnMessage(rtc::Message *pmsg) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001569 switch (pmsg->message_id) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001570 case MSG_EARLYMEDIATIMEOUT:
1571 HandleEarlyMediaTimeout();
1572 break;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001573 case MSG_CHANNEL_ERROR: {
1574 VoiceChannelErrorMessageData* data =
1575 static_cast<VoiceChannelErrorMessageData*>(pmsg->pdata);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001576 delete data;
1577 break;
1578 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001579 default:
1580 BaseChannel::OnMessage(pmsg);
1581 break;
1582 }
1583}
1584
1585void VoiceChannel::OnConnectionMonitorUpdate(
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +00001586 ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001587 SignalConnectionMonitor(this, infos);
1588}
1589
1590void VoiceChannel::OnMediaMonitorUpdate(
1591 VoiceMediaChannel* media_channel, const VoiceMediaInfo& info) {
1592 ASSERT(media_channel == this->media_channel());
1593 SignalMediaMonitor(this, info);
1594}
1595
1596void VoiceChannel::OnAudioMonitorUpdate(AudioMonitor* monitor,
1597 const AudioInfo& info) {
1598 SignalAudioMonitor(this, info);
1599}
1600
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -08001601void VoiceChannel::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const {
1602 GetSupportedAudioCryptoSuites(crypto_suites);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001603}
1604
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001605VideoChannel::VideoChannel(rtc::Thread* thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001606 VideoMediaChannel* media_channel,
deadbeefcbecd352015-09-23 11:50:27 -07001607 TransportController* transport_controller,
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001608 const std::string& content_name,
Fredrik Solenberg7fb711f2015-04-22 15:30:51 +02001609 bool rtcp)
deadbeefcbecd352015-09-23 11:50:27 -07001610 : BaseChannel(thread,
1611 media_channel,
1612 transport_controller,
1613 content_name,
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001614 rtcp),
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001615 renderer_(NULL),
deadbeefcbecd352015-09-23 11:50:27 -07001616 previous_we_(rtc::WE_CLOSE) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001617
1618bool VideoChannel::Init() {
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +00001619 if (!BaseChannel::Init()) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001620 return false;
1621 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001622 return true;
1623}
1624
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001625VideoChannel::~VideoChannel() {
Peter Boström0c4e06b2015-10-07 12:23:21 +02001626 std::vector<uint32_t> screencast_ssrcs;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001627 ScreencastMap::iterator iter;
1628 while (!screencast_capturers_.empty()) {
1629 if (!RemoveScreencast(screencast_capturers_.begin()->first)) {
1630 LOG(LS_ERROR) << "Unable to delete screencast with ssrc "
1631 << screencast_capturers_.begin()->first;
1632 ASSERT(false);
1633 break;
1634 }
1635 }
1636
1637 StopMediaMonitor();
1638 // this can't be done in the base class, since it calls a virtual
1639 DisableMedia_w();
wu@webrtc.org78187522013-10-07 23:32:02 +00001640
1641 Deinit();
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001642}
1643
Peter Boström0c4e06b2015-10-07 12:23:21 +02001644bool VideoChannel::SetRenderer(uint32_t ssrc, VideoRenderer* renderer) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001645 worker_thread()->Invoke<void>(Bind(
1646 &VideoMediaChannel::SetRenderer, media_channel(), ssrc, renderer));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001647 return true;
1648}
1649
1650bool VideoChannel::ApplyViewRequest(const ViewRequest& request) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001651 return InvokeOnWorker(Bind(&VideoChannel::ApplyViewRequest_w, this, request));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001652}
1653
Peter Boström0c4e06b2015-10-07 12:23:21 +02001654bool VideoChannel::AddScreencast(uint32_t ssrc, VideoCapturer* capturer) {
buildbot@webrtc.org65b98d12014-08-07 22:09:08 +00001655 return worker_thread()->Invoke<bool>(Bind(
1656 &VideoChannel::AddScreencast_w, this, ssrc, capturer));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001657}
1658
Peter Boström0c4e06b2015-10-07 12:23:21 +02001659bool VideoChannel::SetCapturer(uint32_t ssrc, VideoCapturer* capturer) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001660 return InvokeOnWorker(Bind(&VideoMediaChannel::SetCapturer,
1661 media_channel(), ssrc, capturer));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001662}
1663
Peter Boström0c4e06b2015-10-07 12:23:21 +02001664bool VideoChannel::RemoveScreencast(uint32_t ssrc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001665 return InvokeOnWorker(Bind(&VideoChannel::RemoveScreencast_w, this, ssrc));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001666}
1667
1668bool VideoChannel::IsScreencasting() {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001669 return InvokeOnWorker(Bind(&VideoChannel::IsScreencasting_w, this));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001670}
1671
Peter Boström0c4e06b2015-10-07 12:23:21 +02001672int VideoChannel::GetScreencastFps(uint32_t ssrc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001673 ScreencastDetailsData data(ssrc);
1674 worker_thread()->Invoke<void>(Bind(
1675 &VideoChannel::GetScreencastDetails_w, this, &data));
wu@webrtc.orgcadf9042013-08-30 21:24:16 +00001676 return data.fps;
1677}
1678
Peter Boström0c4e06b2015-10-07 12:23:21 +02001679int VideoChannel::GetScreencastMaxPixels(uint32_t ssrc) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001680 ScreencastDetailsData data(ssrc);
1681 worker_thread()->Invoke<void>(Bind(
1682 &VideoChannel::GetScreencastDetails_w, this, &data));
wu@webrtc.orgcadf9042013-08-30 21:24:16 +00001683 return data.screencast_max_pixels;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001684}
1685
1686bool VideoChannel::SendIntraFrame() {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001687 worker_thread()->Invoke<void>(Bind(
1688 &VideoMediaChannel::SendIntraFrame, media_channel()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001689 return true;
1690}
1691
1692bool VideoChannel::RequestIntraFrame() {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001693 worker_thread()->Invoke<void>(Bind(
1694 &VideoMediaChannel::RequestIntraFrame, media_channel()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001695 return true;
1696}
1697
Peter Boström0c4e06b2015-10-07 12:23:21 +02001698bool VideoChannel::SetVideoSend(uint32_t ssrc,
deadbeefcbecd352015-09-23 11:50:27 -07001699 bool mute,
solenberg1dd98f32015-09-10 01:57:14 -07001700 const VideoOptions* options) {
deadbeefcbecd352015-09-23 11:50:27 -07001701 return InvokeOnWorker(Bind(&VideoMediaChannel::SetVideoSend, media_channel(),
1702 ssrc, mute, options));
solenberg1dd98f32015-09-10 01:57:14 -07001703}
1704
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001705void VideoChannel::ChangeState() {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001706 // Send outgoing data if we're the active call, we have the remote content,
1707 // and we have had some form of connectivity.
1708 bool send = IsReadyToSend();
1709 if (!media_channel()->SetSend(send)) {
1710 LOG(LS_ERROR) << "Failed to SetSend on video channel";
1711 // TODO(gangji): Report error back to server.
1712 }
1713
Peter Boström34fbfff2015-09-24 19:20:30 +02001714 LOG(LS_INFO) << "Changing video state, send=" << send;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001715}
1716
pbos@webrtc.org058b1f12015-03-04 08:54:32 +00001717bool VideoChannel::GetStats(VideoMediaInfo* stats) {
1718 return InvokeOnWorker(
1719 Bind(&VideoMediaChannel::GetStats, media_channel(), stats));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001720}
1721
1722void VideoChannel::StartMediaMonitor(int cms) {
1723 media_monitor_.reset(new VideoMediaMonitor(media_channel(), worker_thread(),
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001724 rtc::Thread::Current()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001725 media_monitor_->SignalUpdate.connect(
1726 this, &VideoChannel::OnMediaMonitorUpdate);
1727 media_monitor_->Start(cms);
1728}
1729
1730void VideoChannel::StopMediaMonitor() {
1731 if (media_monitor_) {
1732 media_monitor_->Stop();
1733 media_monitor_.reset();
1734 }
1735}
1736
1737const ContentInfo* VideoChannel::GetFirstContent(
1738 const SessionDescription* sdesc) {
1739 return GetFirstVideoContent(sdesc);
1740}
1741
1742bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001743 ContentAction action,
1744 std::string* error_desc) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001745 ASSERT(worker_thread() == rtc::Thread::Current());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001746 LOG(LS_INFO) << "Setting local video description";
1747
1748 const VideoContentDescription* video =
1749 static_cast<const VideoContentDescription*>(content);
1750 ASSERT(video != NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001751 if (!video) {
1752 SafeSetError("Can't find video content in local description.", error_desc);
1753 return false;
1754 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001755
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001756 if (!SetRtpTransportParameters_w(content, action, CS_LOCAL, error_desc)) {
1757 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001758 }
1759
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001760 VideoRecvParameters recv_params = last_recv_params_;
1761 RtpParametersFromMediaDescription(video, &recv_params);
1762 if (!media_channel()->SetRecvParameters(recv_params)) {
1763 SafeSetError("Failed to set local video description recv parameters.",
1764 error_desc);
1765 return false;
1766 }
1767 for (const VideoCodec& codec : video->codecs()) {
1768 bundle_filter()->AddPayloadType(codec.id);
1769 }
1770 last_recv_params_ = recv_params;
1771
1772 // TODO(pthatcher): Move local streams into VideoSendParameters, and
1773 // only give it to the media channel once we have a remote
1774 // description too (without a remote description, we won't be able
1775 // to send them anyway).
1776 if (!UpdateLocalStreams_w(video->streams(), action, error_desc)) {
1777 SafeSetError("Failed to set local video description streams.", error_desc);
1778 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001779 }
1780
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001781 set_local_content_direction(content->direction());
1782 ChangeState();
1783 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001784}
1785
1786bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001787 ContentAction action,
1788 std::string* error_desc) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001789 ASSERT(worker_thread() == rtc::Thread::Current());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001790 LOG(LS_INFO) << "Setting remote video description";
1791
1792 const VideoContentDescription* video =
1793 static_cast<const VideoContentDescription*>(content);
1794 ASSERT(video != NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00001795 if (!video) {
1796 SafeSetError("Can't find video content in remote description.", error_desc);
1797 return false;
1798 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001799
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001800
1801 if (!SetRtpTransportParameters_w(content, action, CS_REMOTE, error_desc)) {
1802 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001803 }
1804
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001805 VideoSendParameters send_params = last_send_params_;
1806 RtpSendParametersFromMediaDescription(video, &send_params);
1807 if (video->conference_mode()) {
Karl Wibergbe579832015-11-10 22:34:18 +01001808 send_params.options.conference_mode = rtc::Optional<bool>(true);
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001809 }
1810 if (!media_channel()->SetSendParameters(send_params)) {
1811 SafeSetError("Failed to set remote video description send parameters.",
1812 error_desc);
1813 return false;
1814 }
1815 last_send_params_ = send_params;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001816
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001817 // TODO(pthatcher): Move remote streams into VideoRecvParameters,
1818 // and only give it to the media channel once we have a local
1819 // description too (without a local description, we won't be able to
1820 // recv them anyway).
1821 if (!UpdateRemoteStreams_w(video->streams(), action, error_desc)) {
1822 SafeSetError("Failed to set remote video description streams.", error_desc);
1823 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001824 }
1825
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001826 if (video->rtp_header_extensions_set()) {
1827 MaybeCacheRtpAbsSendTimeHeaderExtension(video->rtp_header_extensions());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001828 }
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07001829
1830 set_remote_content_direction(content->direction());
1831 ChangeState();
1832 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001833}
1834
1835bool VideoChannel::ApplyViewRequest_w(const ViewRequest& request) {
1836 bool ret = true;
1837 // Set the send format for each of the local streams. If the view request
1838 // does not contain a local stream, set its send format to 0x0, which will
1839 // drop all frames.
1840 for (std::vector<StreamParams>::const_iterator it = local_streams().begin();
1841 it != local_streams().end(); ++it) {
1842 VideoFormat format(0, 0, 0, cricket::FOURCC_I420);
1843 StaticVideoViews::const_iterator view;
1844 for (view = request.static_video_views.begin();
1845 view != request.static_video_views.end(); ++view) {
1846 if (view->selector.Matches(*it)) {
1847 format.width = view->width;
1848 format.height = view->height;
1849 format.interval = cricket::VideoFormat::FpsToInterval(view->framerate);
1850 break;
1851 }
1852 }
1853
1854 ret &= media_channel()->SetSendStreamFormat(it->first_ssrc(), format);
1855 }
1856
1857 // Check if the view request has invalid streams.
1858 for (StaticVideoViews::const_iterator it = request.static_video_views.begin();
1859 it != request.static_video_views.end(); ++it) {
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +00001860 if (!GetStream(local_streams(), it->selector)) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001861 LOG(LS_WARNING) << "View request for ("
1862 << it->selector.ssrc << ", '"
1863 << it->selector.groupid << "', '"
1864 << it->selector.streamid << "'"
1865 << ") is not in the local streams.";
1866 }
1867 }
1868
1869 return ret;
1870}
1871
Peter Boström0c4e06b2015-10-07 12:23:21 +02001872bool VideoChannel::AddScreencast_w(uint32_t ssrc, VideoCapturer* capturer) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001873 if (screencast_capturers_.find(ssrc) != screencast_capturers_.end()) {
buildbot@webrtc.org65b98d12014-08-07 22:09:08 +00001874 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001875 }
buildbot@webrtc.org65b98d12014-08-07 22:09:08 +00001876 capturer->SignalStateChange.connect(this, &VideoChannel::OnStateChange);
1877 screencast_capturers_[ssrc] = capturer;
1878 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001879}
1880
Peter Boström0c4e06b2015-10-07 12:23:21 +02001881bool VideoChannel::RemoveScreencast_w(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001882 ScreencastMap::iterator iter = screencast_capturers_.find(ssrc);
1883 if (iter == screencast_capturers_.end()) {
1884 return false;
1885 }
1886 // Clean up VideoCapturer.
1887 delete iter->second;
1888 screencast_capturers_.erase(iter);
1889 return true;
1890}
1891
1892bool VideoChannel::IsScreencasting_w() const {
1893 return !screencast_capturers_.empty();
1894}
1895
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001896void VideoChannel::GetScreencastDetails_w(
1897 ScreencastDetailsData* data) const {
wu@webrtc.orgcadf9042013-08-30 21:24:16 +00001898 ScreencastMap::const_iterator iter = screencast_capturers_.find(data->ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001899 if (iter == screencast_capturers_.end()) {
wu@webrtc.orgcadf9042013-08-30 21:24:16 +00001900 return;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001901 }
1902 VideoCapturer* capturer = iter->second;
1903 const VideoFormat* video_format = capturer->GetCaptureFormat();
wu@webrtc.orgcadf9042013-08-30 21:24:16 +00001904 data->fps = VideoFormat::IntervalToFps(video_format->interval);
1905 data->screencast_max_pixels = capturer->screencast_max_pixels();
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001906}
1907
Peter Boström0c4e06b2015-10-07 12:23:21 +02001908void VideoChannel::OnScreencastWindowEvent_s(uint32_t ssrc,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001909 rtc::WindowEvent we) {
1910 ASSERT(signaling_thread() == rtc::Thread::Current());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001911 SignalScreencastWindowEvent(ssrc, we);
1912}
1913
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001914void VideoChannel::OnMessage(rtc::Message *pmsg) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001915 switch (pmsg->message_id) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001916 case MSG_SCREENCASTWINDOWEVENT: {
1917 const ScreencastEventMessageData* data =
1918 static_cast<ScreencastEventMessageData*>(pmsg->pdata);
1919 OnScreencastWindowEvent_s(data->ssrc, data->event);
1920 delete data;
1921 break;
1922 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001923 case MSG_CHANNEL_ERROR: {
1924 const VideoChannelErrorMessageData* data =
1925 static_cast<VideoChannelErrorMessageData*>(pmsg->pdata);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001926 delete data;
1927 break;
1928 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001929 default:
1930 BaseChannel::OnMessage(pmsg);
1931 break;
1932 }
1933}
1934
1935void VideoChannel::OnConnectionMonitorUpdate(
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +00001936 ConnectionMonitor* monitor, const std::vector<ConnectionInfo> &infos) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001937 SignalConnectionMonitor(this, infos);
1938}
1939
1940// TODO(pthatcher): Look into removing duplicate code between
1941// audio, video, and data, perhaps by using templates.
1942void VideoChannel::OnMediaMonitorUpdate(
1943 VideoMediaChannel* media_channel, const VideoMediaInfo &info) {
1944 ASSERT(media_channel == this->media_channel());
1945 SignalMediaMonitor(this, info);
1946}
1947
Peter Boström0c4e06b2015-10-07 12:23:21 +02001948void VideoChannel::OnScreencastWindowEvent(uint32_t ssrc,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001949 rtc::WindowEvent event) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001950 ScreencastEventMessageData* pdata =
1951 new ScreencastEventMessageData(ssrc, event);
1952 signaling_thread()->Post(this, MSG_SCREENCASTWINDOWEVENT, pdata);
1953}
1954
1955void VideoChannel::OnStateChange(VideoCapturer* capturer, CaptureState ev) {
1956 // Map capturer events to window events. In the future we may want to simply
1957 // pass these events up directly.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001958 rtc::WindowEvent we;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001959 if (ev == CS_STOPPED) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001960 we = rtc::WE_CLOSE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001961 } else if (ev == CS_PAUSED) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001962 we = rtc::WE_MINIMIZE;
1963 } else if (ev == CS_RUNNING && previous_we_ == rtc::WE_MINIMIZE) {
1964 we = rtc::WE_RESTORE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001965 } else {
1966 return;
1967 }
1968 previous_we_ = we;
1969
Peter Boström0c4e06b2015-10-07 12:23:21 +02001970 uint32_t ssrc = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001971 if (!GetLocalSsrc(capturer, &ssrc)) {
1972 return;
1973 }
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00001974
1975 OnScreencastWindowEvent(ssrc, we);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001976}
1977
Peter Boström0c4e06b2015-10-07 12:23:21 +02001978bool VideoChannel::GetLocalSsrc(const VideoCapturer* capturer, uint32_t* ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001979 *ssrc = 0;
1980 for (ScreencastMap::iterator iter = screencast_capturers_.begin();
1981 iter != screencast_capturers_.end(); ++iter) {
1982 if (iter->second == capturer) {
1983 *ssrc = iter->first;
1984 return true;
1985 }
1986 }
1987 return false;
1988}
1989
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -08001990void VideoChannel::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const {
1991 GetSupportedVideoCryptoSuites(crypto_suites);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001992}
1993
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001994DataChannel::DataChannel(rtc::Thread* thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001995 DataMediaChannel* media_channel,
deadbeefcbecd352015-09-23 11:50:27 -07001996 TransportController* transport_controller,
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001997 const std::string& content_name,
1998 bool rtcp)
deadbeefcbecd352015-09-23 11:50:27 -07001999 : BaseChannel(thread,
2000 media_channel,
2001 transport_controller,
2002 content_name,
2003 rtcp),
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +00002004 data_channel_type_(cricket::DCT_NONE),
deadbeefcbecd352015-09-23 11:50:27 -07002005 ready_to_send_data_(false) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002006
2007DataChannel::~DataChannel() {
2008 StopMediaMonitor();
2009 // this can't be done in the base class, since it calls a virtual
2010 DisableMedia_w();
wu@webrtc.org78187522013-10-07 23:32:02 +00002011
2012 Deinit();
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002013}
2014
2015bool DataChannel::Init() {
pthatcher@webrtc.org6ad507a2015-03-16 20:19:12 +00002016 if (!BaseChannel::Init()) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002017 return false;
2018 }
2019 media_channel()->SignalDataReceived.connect(
2020 this, &DataChannel::OnDataReceived);
wu@webrtc.orgd64719d2013-08-01 00:00:07 +00002021 media_channel()->SignalReadyToSend.connect(
2022 this, &DataChannel::OnDataChannelReadyToSend);
buildbot@webrtc.org1d66be22014-05-29 22:54:24 +00002023 media_channel()->SignalStreamClosedRemotely.connect(
2024 this, &DataChannel::OnStreamClosedRemotely);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002025 return true;
2026}
2027
2028bool DataChannel::SendData(const SendDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00002029 const rtc::Buffer& payload,
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002030 SendDataResult* result) {
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00002031 return InvokeOnWorker(Bind(&DataMediaChannel::SendData,
2032 media_channel(), params, payload, result));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002033}
2034
2035const ContentInfo* DataChannel::GetFirstContent(
2036 const SessionDescription* sdesc) {
2037 return GetFirstDataContent(sdesc);
2038}
2039
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00002040bool DataChannel::WantsPacket(bool rtcp, rtc::Buffer* packet) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002041 if (data_channel_type_ == DCT_SCTP) {
2042 // TODO(pthatcher): Do this in a more robust way by checking for
2043 // SCTP or DTLS.
kwiberg@webrtc.orgeebcab52015-03-24 09:19:06 +00002044 return !IsRtpPacket(packet->data(), packet->size());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002045 } else if (data_channel_type_ == DCT_RTP) {
2046 return BaseChannel::WantsPacket(rtcp, packet);
2047 }
2048 return false;
2049}
2050
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002051bool DataChannel::SetDataChannelType(DataChannelType new_data_channel_type,
2052 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002053 // It hasn't been set before, so set it now.
2054 if (data_channel_type_ == DCT_NONE) {
2055 data_channel_type_ = new_data_channel_type;
2056 return true;
2057 }
2058
2059 // It's been set before, but doesn't match. That's bad.
2060 if (data_channel_type_ != new_data_channel_type) {
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002061 std::ostringstream desc;
2062 desc << "Data channel type mismatch."
2063 << " Expected " << data_channel_type_
2064 << " Got " << new_data_channel_type;
2065 SafeSetError(desc.str(), error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002066 return false;
2067 }
2068
2069 // It's hasn't changed. Nothing to do.
2070 return true;
2071}
2072
2073bool DataChannel::SetDataChannelTypeFromContent(
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002074 const DataContentDescription* content,
2075 std::string* error_desc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002076 bool is_sctp = ((content->protocol() == kMediaProtocolSctp) ||
2077 (content->protocol() == kMediaProtocolDtlsSctp));
2078 DataChannelType data_channel_type = is_sctp ? DCT_SCTP : DCT_RTP;
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002079 return SetDataChannelType(data_channel_type, error_desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002080}
2081
2082bool DataChannel::SetLocalContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002083 ContentAction action,
2084 std::string* error_desc) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00002085 ASSERT(worker_thread() == rtc::Thread::Current());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002086 LOG(LS_INFO) << "Setting local data description";
2087
2088 const DataContentDescription* data =
2089 static_cast<const DataContentDescription*>(content);
2090 ASSERT(data != NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002091 if (!data) {
2092 SafeSetError("Can't find data content in local description.", error_desc);
2093 return false;
2094 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002095
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002096 if (!SetDataChannelTypeFromContent(data, error_desc)) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002097 return false;
2098 }
2099
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07002100 if (data_channel_type_ == DCT_RTP) {
2101 if (!SetRtpTransportParameters_w(content, action, CS_LOCAL, error_desc)) {
2102 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002103 }
2104 }
2105
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07002106 // FYI: We send the SCTP port number (not to be confused with the
2107 // underlying UDP port number) as a codec parameter. So even SCTP
2108 // data channels need codecs.
2109 DataRecvParameters recv_params = last_recv_params_;
2110 RtpParametersFromMediaDescription(data, &recv_params);
2111 if (!media_channel()->SetRecvParameters(recv_params)) {
2112 SafeSetError("Failed to set remote data description recv parameters.",
2113 error_desc);
2114 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002115 }
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07002116 if (data_channel_type_ == DCT_RTP) {
2117 for (const DataCodec& codec : data->codecs()) {
2118 bundle_filter()->AddPayloadType(codec.id);
2119 }
2120 }
2121 last_recv_params_ = recv_params;
2122
2123 // TODO(pthatcher): Move local streams into DataSendParameters, and
2124 // only give it to the media channel once we have a remote
2125 // description too (without a remote description, we won't be able
2126 // to send them anyway).
2127 if (!UpdateLocalStreams_w(data->streams(), action, error_desc)) {
2128 SafeSetError("Failed to set local data description streams.", error_desc);
2129 return false;
2130 }
2131
2132 set_local_content_direction(content->direction());
2133 ChangeState();
2134 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002135}
2136
2137bool DataChannel::SetRemoteContent_w(const MediaContentDescription* content,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002138 ContentAction action,
2139 std::string* error_desc) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00002140 ASSERT(worker_thread() == rtc::Thread::Current());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002141
2142 const DataContentDescription* data =
2143 static_cast<const DataContentDescription*>(content);
2144 ASSERT(data != NULL);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002145 if (!data) {
2146 SafeSetError("Can't find data content in remote description.", error_desc);
2147 return false;
2148 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002149
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07002150 // If the remote data doesn't have codecs and isn't an update, it
2151 // must be empty, so ignore it.
2152 if (!data->has_codecs() && action != CA_UPDATE) {
2153 return true;
2154 }
2155
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +00002156 if (!SetDataChannelTypeFromContent(data, error_desc)) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002157 return false;
2158 }
2159
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07002160 LOG(LS_INFO) << "Setting remote data description";
2161 if (data_channel_type_ == DCT_RTP &&
2162 !SetRtpTransportParameters_w(content, action, CS_REMOTE, error_desc)) {
2163 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002164 }
2165
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07002166
2167 DataSendParameters send_params = last_send_params_;
2168 RtpSendParametersFromMediaDescription<DataCodec>(data, &send_params);
2169 if (!media_channel()->SetSendParameters(send_params)) {
2170 SafeSetError("Failed to set remote data description send parameters.",
2171 error_desc);
2172 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002173 }
Peter Thatcherc2ee2c82015-08-07 16:05:34 -07002174 last_send_params_ = send_params;
2175
2176 // TODO(pthatcher): Move remote streams into DataRecvParameters,
2177 // and only give it to the media channel once we have a local
2178 // description too (without a local description, we won't be able to
2179 // recv them anyway).
2180 if (!UpdateRemoteStreams_w(data->streams(), action, error_desc)) {
2181 SafeSetError("Failed to set remote data description streams.",
2182 error_desc);
2183 return false;
2184 }
2185
2186 set_remote_content_direction(content->direction());
2187 ChangeState();
2188 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002189}
2190
2191void DataChannel::ChangeState() {
2192 // Render incoming data if we're the active call, and we have the local
2193 // content. We receive data on the default channel and multiplexed streams.
2194 bool recv = IsReadyToReceive();
2195 if (!media_channel()->SetReceive(recv)) {
2196 LOG(LS_ERROR) << "Failed to SetReceive on data channel";
2197 }
2198
2199 // Send outgoing data if we're the active call, we have the remote content,
2200 // and we have had some form of connectivity.
2201 bool send = IsReadyToSend();
2202 if (!media_channel()->SetSend(send)) {
2203 LOG(LS_ERROR) << "Failed to SetSend on data channel";
2204 }
2205
sergeyu@chromium.org9cf037b2014-02-07 19:03:26 +00002206 // Trigger SignalReadyToSendData asynchronously.
2207 OnDataChannelReadyToSend(send);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002208
2209 LOG(LS_INFO) << "Changing data state, recv=" << recv << " send=" << send;
2210}
2211
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00002212void DataChannel::OnMessage(rtc::Message *pmsg) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002213 switch (pmsg->message_id) {
2214 case MSG_READYTOSENDDATA: {
wu@webrtc.orgd64719d2013-08-01 00:00:07 +00002215 DataChannelReadyToSendMessageData* data =
2216 static_cast<DataChannelReadyToSendMessageData*>(pmsg->pdata);
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +00002217 ready_to_send_data_ = data->data();
2218 SignalReadyToSendData(ready_to_send_data_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002219 delete data;
2220 break;
2221 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002222 case MSG_DATARECEIVED: {
2223 DataReceivedMessageData* data =
2224 static_cast<DataReceivedMessageData*>(pmsg->pdata);
2225 SignalDataReceived(this, data->params, data->payload);
2226 delete data;
2227 break;
2228 }
2229 case MSG_CHANNEL_ERROR: {
2230 const DataChannelErrorMessageData* data =
2231 static_cast<DataChannelErrorMessageData*>(pmsg->pdata);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002232 delete data;
2233 break;
2234 }
buildbot@webrtc.org1d66be22014-05-29 22:54:24 +00002235 case MSG_STREAMCLOSEDREMOTELY: {
Peter Boström0c4e06b2015-10-07 12:23:21 +02002236 rtc::TypedMessageData<uint32_t>* data =
2237 static_cast<rtc::TypedMessageData<uint32_t>*>(pmsg->pdata);
buildbot@webrtc.org1d66be22014-05-29 22:54:24 +00002238 SignalStreamClosedRemotely(data->data());
2239 delete data;
2240 break;
2241 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002242 default:
2243 BaseChannel::OnMessage(pmsg);
2244 break;
2245 }
2246}
2247
2248void DataChannel::OnConnectionMonitorUpdate(
pthatcher@webrtc.orgb4aac132015-03-13 18:25:21 +00002249 ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002250 SignalConnectionMonitor(this, infos);
2251}
2252
2253void DataChannel::StartMediaMonitor(int cms) {
2254 media_monitor_.reset(new DataMediaMonitor(media_channel(), worker_thread(),
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00002255 rtc::Thread::Current()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002256 media_monitor_->SignalUpdate.connect(
2257 this, &DataChannel::OnMediaMonitorUpdate);
2258 media_monitor_->Start(cms);
2259}
2260
2261void DataChannel::StopMediaMonitor() {
2262 if (media_monitor_) {
2263 media_monitor_->Stop();
2264 media_monitor_->SignalUpdate.disconnect(this);
2265 media_monitor_.reset();
2266 }
2267}
2268
2269void DataChannel::OnMediaMonitorUpdate(
2270 DataMediaChannel* media_channel, const DataMediaInfo& info) {
2271 ASSERT(media_channel == this->media_channel());
2272 SignalMediaMonitor(this, info);
2273}
2274
2275void DataChannel::OnDataReceived(
2276 const ReceiveDataParams& params, const char* data, size_t len) {
2277 DataReceivedMessageData* msg = new DataReceivedMessageData(
2278 params, data, len);
2279 signaling_thread()->Post(this, MSG_DATARECEIVED, msg);
2280}
2281
Peter Boström0c4e06b2015-10-07 12:23:21 +02002282void DataChannel::OnDataChannelError(uint32_t ssrc,
2283 DataMediaChannel::Error err) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002284 DataChannelErrorMessageData* data = new DataChannelErrorMessageData(
2285 ssrc, err);
2286 signaling_thread()->Post(this, MSG_CHANNEL_ERROR, data);
2287}
2288
wu@webrtc.orgd64719d2013-08-01 00:00:07 +00002289void DataChannel::OnDataChannelReadyToSend(bool writable) {
2290 // This is usded for congestion control to indicate that the stream is ready
2291 // to send by the MediaChannel, as opposed to OnReadyToSend, which indicates
2292 // that the transport channel is ready.
2293 signaling_thread()->Post(this, MSG_READYTOSENDDATA,
2294 new DataChannelReadyToSendMessageData(writable));
2295}
2296
Guo-wei Shieh521ed7b2015-11-18 19:41:53 -08002297void DataChannel::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const {
2298 GetSupportedDataCryptoSuites(crypto_suites);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002299}
2300
2301bool DataChannel::ShouldSetupDtlsSrtp() const {
Guo-wei Shieh9c38c2d2015-12-05 09:46:07 -08002302 return (data_channel_type_ == DCT_RTP) && BaseChannel::ShouldSetupDtlsSrtp();
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002303}
2304
Peter Boström0c4e06b2015-10-07 12:23:21 +02002305void DataChannel::OnStreamClosedRemotely(uint32_t sid) {
2306 rtc::TypedMessageData<uint32_t>* message =
2307 new rtc::TypedMessageData<uint32_t>(sid);
buildbot@webrtc.org1d66be22014-05-29 22:54:24 +00002308 signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message);
2309}
2310
henrike@webrtc.org28e20752013-07-10 00:45:36 +00002311} // namespace cricket